Computer Networks I

A Book Of
COMPUTER
NETWORKS-I
For S.Y.B.Sc. Computer Science : Semester – IV (Paper – II)
[Course Code CS 242 : Credits - 2]
CBCS Pattern
As Per New Syllabus, Effective from June 2020
Dr. Ms. Manisha Bharambe Mrs. Veena K. Gandhi

M.Sc. (Comp. Sci.), M.Phil. Ph.D. (Comp. Sci.) M.C.S., M.Phil (Comp. Sci.), UGC-NET
Vice Principal, Associate Professor, Dept. of Comp. Sci., Head, Department of B.C.A. (Science),
MES's Abasaheb Garware College, Abeda Inamdar Senior College,
Pune Pune
Price ` 330.00
N5542
COMPUTER NETWORKS-I ISBN 978-93-90506-39-2
First Edition : January 2021
© : Authors
The text of this publication, or any part thereof, should not be reproduced or transmitted in any form or stored in any computer
storage system or device for distribution including photocopy, recording, taping or information retrieval system or reproduced on any disc,
tape, perforated media or other information storage device etc., without the written permission of Authors with whom the rights are
reserved. Breach of this condition is liable for legal action.
Every effort has been made to avoid errors or omissions in this publication. In spite of this, errors may have crept in. Any mistake, error
or discrepancy so noted and shall be brought to our notice shall be taken care of in the next edition. It is notified that neither the publisher
nor the authors or seller shall be responsible for any damage or loss of action to any one, of any kind, in any manner, therefrom.
Published By : Polyplate Printed By :

NIRALI PRAKASHAN YOGIRAJ PRINTERS AND BINDERS
Abhyudaya Pragati, 1312, Shivaji Nagar, Survey No. 10/1A, Ghule Industrial Estate
Off J.M. Road, Pune – 411005 Nanded Gaon Road
Tel - (020) 25512336/37/39, Fax - (020) 25511379 Nanded, Pune - 411041
Email : [email protected] Mobile No. 9404233041/9850046517
DISTRIBUTION CENTRES
PUNE
Nirali Prakashan : 119, Budhwar Peth, Jogeshwari Mandir Lane, Pune 411002, Maharashtra
(For orders within Pune) Tel : (020) 2445 2044; Mobile : 9657703145
Email : [email protected]
Nirali Prakashan : S. No. 28/27, Dhayari, Near Asian College Pune 411041
(For orders outside Pune) Tel : (020) 24690204; Mobile : 9657703143
MUMBAI
Nirali Prakashan : 385, S.V.P. Road, Rasdhara Co-op. Hsg. Society Ltd.,
Girgaum, Mumbai 400004, Maharashtra; Mobile : 9320129587
Tel : (022) 2385 6339 / 2386 9976, Fax : (022) 2386 9976
DISTRIBUTION BRANCHES
JALGAON
Nirali Prakashan : 34, V. V. Golani Market, Navi Peth, Jalgaon 425001, Maharashtra,
Tel : (0257) 222 0395, Mob : 94234 91860; Email : [email protected]
KOLHAPUR
Nirali Prakashan : New Mahadvar Road, Kedar Plaza, 1st Floor Opp. IDBI Bank, Kolhapur 416 012
Maharashtra. Mob : 9850046155; Email : [email protected]
NAGPUR
Nirali Prakashan : Above Maratha Mandir, Shop No. 3, First Floor,
Rani Jhanshi Square, Sitabuldi, Nagpur 440012, Maharashtra
Tel : (0712) 254 7129; Email : [email protected]
DELHI
Nirali Prakashan : 4593/15, Basement, Agarwal Lane, Ansari Road, Daryaganj
Near Times of India Building, New Delhi 110002 Mob : 08505972553
BENGALURU
Nirali Prakashan : Maitri Ground Floor, Jaya Apartments, No. 99, 6th Cross, 6th Main,
Malleswaram, Bengaluru 560003, Karnataka; Mob : 9449043034
Email: [email protected]
Other Branches : Hyderabad, Chennai
Note : Every possible effort has been made to avoid errors or omissions in this book. In spite this, errors may have crept in. Any type of
error or mistake so noted, and shall be brought to our notice, shall be taken care of in the next edition. It is notified that neither the publisher,
nor the author or book seller shall be responsible for any damage or loss of action to any one of any kind, in any manner, therefrom. The
reader must cross check all the facts and contents with original Government notification or publications.
[email protected] | www.pragationline.com
Also find us on www.facebook.com/niralibooks
Preface …
We take an opportunity to present this Text Book on "Computer Networks-I" to the

students of Second Year B. Sc. (Computer Science) Semester-IV as per the New Syllabus,
June 2020.
The book has its own unique features. It brings out the subject in a very simple and lucid
manner for easy and comprehensive understanding of the basic concepts. The book covers
theory of Introduction to Networks and Network Models, Lower Layers, Network Layer and
Transport Layer.
A special word of thank to Shri. Dineshbhai Furia, and Mr. Jignesh Furia for
showing full faith in us to write this text book. We also thank to Mr. Amar Salunkhe and
Mrs. Prachi Sawant of M/s Nirali Prakashan for their excellent co-operation.
We also thank Mr. Ravindra Walodare, Mr. Sachin Shinde, Mr. Ashok Bodke,
Mr. Moshin Sayyed and Mr. Nitin Thorat.
Although every care has been taken to check mistakes and misprints, any errors,
omission and suggestions from teachers and students for the improvement of this text book
shall be most welcome.
Authors
Syllabus …
1. Introduction to Networks and Network Models (4 Hrs.)
1.1 Data Communication, Components, Data Representation
1.2 Networks, Network Criteria, Network Types - LAN, WAN, Switching, The Internet,
Accessing the Internet
1.3 Network Software - Protocol Hierarchies, Design Issues of the Layer, Connection Oriented
and Connectionless Services
1.4 Reference Models - OSI Reference Models, TCP/IP Reference Model, Connection Devices
in different Layers, Comparison of OSI and TCP/IP Reference Models
2. Lower Layers (10 Hrs.)
2.1 Communication at the Physical Layer, Data Rate Limits - Noiseless Channel (Nyquist Bit
Rate), Noisy Channel (Shannon Capacity), Performance - Bandwidth, Throughput, Latency,
Bandwidth-Delay Product, Jitter
2.2 Design Issues of Data Link Layer, Services - Framing, Flow Control, Error Control,
Congestion Control, Link Layer Addressing
2.3 Framing Methods - Character Count, Flag Bytes with Byte Stuffing, Flags Bits with Bit
Stuffing, Physical Layer Coding Violations
2.4 The Channel Allocation Problem, Static and Dynamic Allocation, Media Access Methods -
Taxonomy of Multiple - Access Protocols
2.5 Switching and TCP/IP Layers, Types - Circuit Switching, Packet Switching and Message
Switching
2.6 Wired LANs - Standard Ethernet Characteristics, Addressing, Access Method,
Implementation, Fast and Gigabit Ethernet
2.7 Wireless LANs - Architectural Comparison, Characteristics, Access Control, IEEE 802.11
Architecture, Physical Layer, MAC Sublayer, Bluetooth Architecture, Layers
3. Network Layer (12 Hrs.)
3.1 Network Layer Services - Packetizing, Routing and Forwarding, other Services
3.2 Open and Closed Loop Congestion Control
3.3 IPv4 Addressing - Address Space, Classful Addressing, Subnetting, Supernetting, Classless
Addressing, Network Address Resolution (NAT)
3.4 Forwarding of IP Packets - Based on Destination Address, Based on Label
3.5 Network Layer Protocols - Internet Protocol (IP), IPv4 Datagram Format, Fragmentation,
Options
3.6 Mobile IP - Addressing, Agents, Three Phases
3.7 Next Generation IP - IPv6 Address Representation, Address Space, Address Types, IPv6
Protocol, Packet Format, Extension Header, Difference between IPv4 and IPv6
3.8 Routing - General Idea, Algorithms - Distance Vector Routing, Link State Routing, Path -
Vector Routing
4. Transport Layer (10 Hrs.)
4.1 Transport Layer Services - Process-to-Process Communication, Addressing, Encapsulation
and Decapsulation, Multiplexing and Demultiplexing, Flow Control, Pushing or Pulling,
Flow Control, Buffers, Sequence Numbers, Acknowledgements, Sliding Window,
Congestion Control
4.2 Connectionless and Connection-oriented Service, Port Numbers
4.3 Transport Layer Protocols - User Datagram Protocol, User Datagram, UDP Services
4.4 Transmission Control Protocol - TCP Services, TCP Features, TCP Segment Format, Three -
way Handshake for Connection Establishment and Termination, State Transition Diagram,
Windows in TCP
Contents …
1. Introduction to Networks and Network Models 1.1 – 1.82
2. Lower Layers 2.1 – 2.98
3. Network Layer 3.1 – 3.90
4. Transport Layer 4.1 – 4.48

CHAPTER
1
Introduction to Networks and
Network Models
Objectives…
To understand Concepts in Data Communication
To learn Network and Computer Network
To study Basic Concepts of Internet
To learn Network Software
To study Reference Models (OSI and TCP/IP)
1.0 INTRODUCTION
• Data communications and computer networks are two of the fastest growing
technological areas in today’s modern world. This is because there is an almost
unlimited demand for information transfer.
• The word ‘communication’ is derived from the Latin word ‘communicare’, which
means 'to share'. Communication means the transfer of information between humans,
computers or machines in a meaningful way.
• Communication is the process of establishing connection or link between two entities
for the transfer/exchange of information.
• When we communicate, we share information. This sharing can be local or remote.
Local communication can be face to face between individuals, while remote
communication takes over distance.
• Data communication is a process of exchanging data or information between two
devices (a sender and a receiver) through some kind of transmission/ communication
medium such as coaxial cable or fiber optic cable (wired) and/or air (wireless).
• A group of computers and other devices such as printers, hubs, modems etc.,
connected together is called a network. The interconnected computers can share
resources, which is called networking.
• A computer network provides the facility of information exchange/transfer among the
computers, connected to it.
1.1
Computer Networks – I (S.Y.B.Sc. Comp. Science) Introduction to Networks & Network Models
• A network is a combination of hardware and software that sends data from one point
to another.
o The hardware consists of the physical equipment that carries signals from one
point to the network to another.
o The software consists of instruction sets, that make possible the services that we
expect from a network.
• A reference model in networking is a conceptual layout that describes how
communication between devices should occur.
• For efficient communication, the reference model identifies the tasks involved in
inter-computer communication and divides them in logical groups called layers, with
each layer performing a specific function.
• The purpose of the reference model was to define an architectural framework that
defines the logical communication tasks that are required to move information
between different computer systems.
• In networking the two most common reference models are Open Systems
Interconnection (OSI) reference model and TCP/IP (Transmission Control
Protocol/Internet Protocol) reference model.
• In this chapter we will study basic concepts in data communications and networking,
reference models, network software, Internet and so on.
1.1 DATA COMMUNICATION

• Today, technologies related to data communications and networking may be the
fastest growing in our culture and/or our modern society.
• The appearance of new social networking applications like Facebook, Twitter etc.,
every year is a testimony to this claim.
• People use the Internet more and more every day and they use the Internet for
research, banking, shopping, ticket booking (bus, airplane, railway etc.), weather,
education and so on.
• Communication is the basic process of information exchange. Communication,
whether between human beings or computer systems, involves transfer of
information from a sender to a receiver.
• Communication is the conveyance of a message from one entity, called the source or
transmitter or sender to another entity called the destination or receiver, using a
communication channel/medium.
• Communication is defined as, transfer of information such as thoughts and messages
between two entities. The process of sending or receiving data between two
points/entities of a computer network is known as data communication.
1.2
• Data communication is the exchange of information between two computers capable

of generating processing and interpreting data.
• The data communication process shown in Fig. 1.1.
Fig. 1.1: Communication between Two Computers

• Fig. 1.1 shows communication between one computer (sender/source) sending a
message to another computer (receiver/destination) over a wire or radio waves called
transmission media.
• The data plays an important role in networking. Data is defined as ‘raw facts and
figures before they have been processed’.
• The meaningful, logical and processed data is called information. Information can be
defined as ‘meaningful data or processed data’.
Comparison between Data and Information:
Sr. No. Data Information
1. Data is a collection of raw facts and Information is processed data.
figures.
2. Data is unarranged and Information is arranged and organized
unorganized.
3. Data is a raw (meaningless) facts Information is meaningful.
that required to be processed to
make it meaningful.
4. Data is input. Information is output.
5. Data is raw material for Information is the final product of data.
information.
6. Data depends upon the sources. Information depends upon data.
• Data communication refers to the exchange of data between a source and a receiver
via some form of transmission medium.
• The effectiveness of a data communication systems depends on four fundamental
characteristics as given below:
1. Delivery: The system must deliver data to the correct destination. Data must be
received by the intended device or user.
1.3
2. Accuracy: The system must deliver the data accurately i.e., without any error.
3. Timeliness: The system must deliver data in a timely manner. Data delivered late
is useless.
4. Jitter: Jitter refers to the variation in the packet arrival time. Jitter is the uneven
delay in the delivery of audio or video packets.
1.1.1 Definition of Data Communication

• The term data communication can be defined as “the process of using computing and
communication technologies to transfer data from one place to another, and vice
versa”. OR
• Data communication refers to the exchange of data between a source and a receiver
via form of transmission media such as a wire cable (twisted-pair cable or coaxial
cable) or wireless (air). OR
• “The transfer or exchange of information from one computer to another is known as
data communication”.
1.1.2 Components of Data Communication
• The purpose of data communications is to provide the rules and regulations that allow
computers with different disk operating systems, languages, cabling and locations to
share resources.
• Data communication refers to the exchange of information/data between two devices
through some form of wired or wireless transmission medium.
• Data communication includes the transfer of data, the method of transfer and the
preservation of the data during the transfer/exchange process.
• To initiate data communication, the communicating devices should be a part of a data
communication system that is formed by the collection of physical equipment’s
(hardware) and programs (software).
• A data communication system has five components namely sender, message, receiver,
transmission medium, and protocol are shown in Fig. 1.2.
Step 1:
Step 2: Step 1:
Step 3: Step 2:
Step 3:
Step n Step n
Protocol Message Protocol
Medium
Sender Receiver
Fig. 1.2: Components of Data Communications
1.4
• Fig. 1.2 shows the following components of data communication:

1. Sender: The sender is the device that sends the data message. It can be a
computer, workstation, telephone handset, and so on.
2. Message: The message is the information (data) to be communicated. It can consist
of text, pictures, sound, or video- or any combination of these.
3. Medium: The transmission medium is the physical path by which a message
travels from sender to receiver. It could be a twisted-pair wire, coaxial cable, fiber-
optic cable, or radio waves.
4. Protocol: A protocol is a set of rules that governs data communications. It
represents an agreement between the communicating devices. Without a protocol,
two devices may be connected but could not communicate.
A protocol refers to a set of rules (agreed upon by the sender and the receiver) that
coordinates the exchange of information. A protocol is a set of rules that governs
data communication. Protocol is very important for networking without protocol
communication cannot occur. A protocol is defined as, "a formal set of rules,
conventions and data structure that governs how computers and other network
devices exchange information over a network."
A protocol defines basic elements namely, Syntax (what is to be communicated?)
Semantics (how it is to be communicated?) and Timing (when it should be
communicated?). (April 17, Oct. 18)
5. Receiver: The receiver is the device that receives the message. It can be computer,
workstation, telephone handset, television, and so on.
1.1.3 Data Representation

• Data representation refers to the methods used internally to represent information
stored in a computer.
• Information today comes in various forms like text, numbers, images, audio, and
video.
1. Text: In data communication, text is represented as a bit pattern, a sequence of 0s
or 1s. Different sets of bit patterns have been designed to represent text symbols.
The process of representing symbols is called coding. Today, Unicode (32 bits) and
American Standard Code for Information Interchange [(ASCII) specifies character
values from 0 to 127] used to represent a symbol or character used in any language
in the world.
2. Numbers: Numbers are also represented by bit patterns. ASCII is not used to
represent numbers. the number is directly converted into binary to simplify
mathematical operations.
1.5
3. Images: Images are also represented by bit patterns. An image is composed of a

matrix of pixels (picture elements). Each pixel in an image is a small dot and each
pixel is assigned a bit pattern whose size depends on the nature of the image. To
represent a color image, a method like RGB and YCM are used. RGB method uses
primary colors Red, Green, and Blue to make every color. YCM method
uses colors Yellow, Cyan, and Magenta to make every color.
4. Audio: Audio refers to the recording or broadcasting of sound or music. It is
continuous, not discrete. The sound is recorded with a microphone and then
digitized to represent in the form of bit-patterns.
5. Video: Video refers to the recording or broadcasting of a picture or movie. Video
can either be produced as a continuous entity by a TV camera or video can be a
combination of images, each discrete entity, arranged to convey the idea of
motion.
1.1.4 Data Flow (Oct. 17)
• The direction of data flow between two linked devices is called as mode of
communication.
• There are three types of direction of data flow namely, simplex communication mode,
half-duplex communication mode and full-duplex communication mode.
1. Simplex Communication Mode:
• In simplex mode, the communication can take place in only one direction.
• In simplex mode, a terminal can only send data and cannot receive it or it can only
receive data but cannot send it. It means that in simplex mode communication is
unidirectional.
• Fig. 1.3 shows simplex mode in data communication.
Direction of data
Mainframe Monitor
Fig. 1.3: Simplex Communication Mode
• In computer systems, the keyboard, monitor and printer are examples of simplex
communication. The keyboard can only be used to enter data into the computer, while
monitor and printer can only accept (display/print) output.
Advantages of Simplex Communication Mode:
(i) Very simple and easy communication method.
1.6
(ii) Cheaper in cost.

Disadvantages Simplex Communication Mode:
(i) Only allows for communication in one direction.
(ii) Simplex transmission is not often used because it is not possible to send back error
to the transmit end.
2. Half-Duplex Communication Mode:
• In half-duplex mode, the communication can take place in both directions, but only in
one direction at a time.
• In this mode, data is sent and received alternatively. It is like a one-lane bridge where
two-way traffic must give way in order to cross the other.
• In simple words, in half duplex mode, at a time only one end transmits data while the
other end receives.
• A walkie-talkie operates in half duplex mode. It can only send or receive a
transmission at any given time. It cannot do both at the same time.
• Fig. 1.4 shows half-duplex communication mode.
Direction of data at time 1
Direction of data at time 2
Workstation Workstation
Fig. 1.4: Half-duplex Communication Mode
Advantages Half Duplex Communication Mode:

(i) Enable two-way communication.
(ii) Low cost than full duplex communication mode.
Disadvantages Half Duplex Communication Mode:
(i) Only one device can transmit at a time.
(ii) Higher cost than simplex mode.
3. Full-Duplex Communication Mode:
• In full-duplex mode, the communication can take place in both directions
simultaneously, i.e. at the same time on the same channel.
For example, the telephone communication system is an example of full-duplex
communication mode.
• Fig. 1.5 shows full-duplex communication mode.
1.7
Direction of data all the time 1
Fig. 1.5: Full Duplex Communication Mode
Advantages:
(i) Enables two-way communication simultaneously.
(ii) Fastest method of data communication.
Disadvantages:
(i) More expensive and complex method.
(ii) Two bandwidth channels is required for data transmission.
1.2 NETWORKS
• A network is the interconnection of a set of devices (like a host (desktop, laptop,
workstation, smartphone) and/or connecting devices like router, switch, modem etc.).
• A network is nothing more than two or more computers connected by a cable or by a
wireless radio waves connection so that they can exchange information.
• A communication network is a set of devices connected by channels on links and
provides a service between users located at various geographical points.
• A network is defined as, “an interconnected collection of autonomous computers”.
Two computers are said to be interconnected if they are capable of exchanging
information.
1.2.1 Computer Networks (April 17, 19)
• A computer network is an interconnection of computers and computing devices using

either wires or radio waves over small or large geographical areas.
• A computer network refers to a collection of two or more computers (nodes) which are
connected together to share information and resources.
• A network is the interconnection of a set of computing devices capable of
communication.
• A network is a set of devices, (often referred to as nodes) connected by communication
media links (See Fig. 1.6).
• A node can be a computer, printer, or any other device capable of sending and/or
receiving data generated by other nodes on the network.
• The links connecting the devices are often called communication channels. The term
channel refers to a communication path between two communicating devices.
1.8
Computer 1
Printer
Nodes
Links
Server
Computer 2
Fig. 1.6: Network Nodes and Links

• Computer network is a set or collection of computing devices that are linked to
each other in order to communicate and share their resources with each other.
• In simple words, a computer network is a group of interconnected computing
devices. The interconnected computers can share resources, which is called
networking.
• Computer network is divided into wired and wireless networks. Fig. 1.7 shows
diagrammatic representation of computer networks (wired and wireless).
Fig. 1.7: Computer Network (Wired and Wireless)

1.9
• A wired network is simply, a collection of nodes connected by cables like Ethernet,

coaxial, fiber optic cable etc. A wireless network, which uses high-frequency radio
waves to communicate between nodes.
• The following characteristics should be considered in computer network design and
ongoing maintenance:
1. Availability: Availability is typically measured in a percentage based on the
number of minutes that exist in a year. Therefore, uptime would be the number of
minutes the network is available divided by the number of minutes in a year.
2. Cost: Includes the cost of the network components, their installation, and their
ongoing maintenance.
3. Reliability: Defines the reliability of the network components and the connectivity
between them. Mean Time Between Failures (MTBF) is commonly used to measure
reliability.
4. Security: Includes the protection of the network components and the data they
contain and/or the data transmitted between them.
5. Speed: Includes how fast data is transmitted between network endpoints, (the data
rate).
6. Scalability: Defines how well the network can adapt to new growth, including
new users, applications, and network components.
7. Topology: Describes the physical cabling layout and the logical way data moves
between components.
8. Integration: All the components of the network work in a coordinated manner for
a seamless user experience.
9. Sharing: Computer networks enable sharing of files, software, hardware resources
and computing capabilities.
1.2.2 Definition of Computer Network

• The old model of a single computer serving all of the organization’s computational
needs has been replaced by one in which a large number of separate but
interconnected computers do the job. These systems are called computer networks.
• A computer network, often simply referred to as a network, is a collection of
computers and computing devices connected by communication channels that
facilitate communication among users and allow shared resources.
• A computer network provides the facility of information exchange/transfer among the
computers, connected to it.
• Computer networking is a very important and crucial part of Information Technology
(IT). Millions of computers are networked together to form the Internet.
1.10
• Networking plays an important role in every kind of organization from small to

medium sized, in banks, multinational companies, stock exchanges, airports, hospitals,
police stations, post offices, colleges, universities and even in the home. In short
networking plays an important role everywhere where computers are used.
• A computer network can be defined as "an interconnected collection of autonomous
computers and computing devices". OR
• A computer network is "an interconnection of computers and computing equipment
like printer, scanner etc. using either wires or radio waves (wireless) made to share
hardware and software resources".
• Fig. 1.8 shows a typical computer network.
Server
Network
Network
Users (clients)
Fig. 1.8: Typical Computer Network
• The computer networks are playing an important role in providing services to large or
small or medium organizations as well as to the individual common man.
• Network services are the things that a network can do. The major networking services
are:
1. File Services: This includes file transfer, storage, data migration, file update,
synchronization and achieving.
2. Printing Services: This service provides shared access to valuable printing
devices.
3. Message Services: This service facilitates email, voice mails and coordinate object
oriented applications.
4. Application Services: This services allows us to centralize high profile
applications to increase performance and scalability.
5. Database Services: This involves coordination of distributed data and replication.
1.2.3 Network Criteria

• To be considered effective and efficient, a computer network must meet a number of
criteria. The most important of computer network criteria are performance, security
and reliability.
1.11
1. Performance:
• Performance of computer networks can be measured in many ways, including transit
time (propagation delay) and response time (speed of operation).
• Transit time is the amount of time required for a message to travel from one device to
another in a network. Response time in the network is the elapsed time between an
inquiry and a response.
• The performance of a network depends on a number of factors. Some of them are
explained below:
(i) Type of Transmission Medium:
o The transmission medium defines the speed at which data can travel through a
connection (the data rate). Today's networks are moving to faster and faster
transmission media, such as fiber-optic cabling.
o A transmission medium that can carry data at 100 megabits per second is 10 times
more powerful than a medium that can carry data at only 10 megabits per second.
(ii) Number of Users:
o Having a large number of concurrent users can slow response time in a computer
network not designed to coordinate heavy traffic loads.
o The design of a given computer network is based on an assessment of the average
number of users that will be communicating at any one time.
o When the actual number of users can exceed the average and thereby decrease
performance. How a network responds to loading is a measure of its performance?
(iii) Software:
o The software used to process data at the sender, receiver, and intermediate nodes
also affects network performance.
o Moving a message from node to node through a network requires processing to
transform the raw data into transmittable signals, to route these signals to the
proper destination, to ensure error-free delivery, and to recast the signals into a
form the receiver can use.
o The software that provides these services affects both the speed and the reliability
of a network link. A well designed software can speed the process and make data
transmission more effective and efficient.
(iv) Hardware:
o The types of hardware included in a computer network affect both the speed and
capacity of data transmission. A higher-speed computer with greater storage
capacity provides better performance.
1.12
2. Security:
• Network security issues in a computer network include protecting data from
unauthorized access and viruses.
(i) Unauthorized Access:
o For a computer network to be useful, sensitive data must be protected from
unauthorized access. Protection can be accomplished at a number of levels.
o At the lowest level are user identification codes and passwords while at a higher
level are encryption techniques.
o In these mechanisms, data are systematically altered/modified in such a way that if
they are intercepted by an unauthorized user, they will be unintelligible.
(ii) Viruses:
o A computer network is accessible from many points, it can be susceptible to
computer viruses. A virus is an illicitly introduced code that damages the
computer system.
o A good computer network is protected from viruses by hardware and software
designed specifically for that purpose.
3. Reliability:
• In addition to accuracy of delivery, network reliability of a computer network is
measured by following factors:
(i) Frequency of Failure:
o All networks fail occasionally. A network that fails often, however, is of little value
to a user.
(ii) Recovery Time of a Network After a Failure:
o How long does it take to restore service? A computer network that recovers
quickly is more useful than one that does not.
(iii) Catastrophe:
o Computer networks must be protected from catastrophic events such as fire,
earthquake or theft.
o One protection against unforeseen damage is a reliable system to back up
computer network software.
1.2.4 Physical Structures

• Before studying computer networks, we need to define some network attributes like
type of connection and physical topology.
1.13
1.2.4.1 Type of Connection

• In order to communicate to each other, two or more devices in a computer network,
must be connected using a link.
• A link is a communications pathway that transfers data from one device to another.
For communication to occurs, two devices must be connected in some way to the same
link at the same time.
• The two types of connections are point-to-point and multipoint. Both types of
connections describe a method to connect two or more communication devices in a
link.
1. Point-to-Point Connection: (April 16)
• Connection between two directly interconnected devices/nodes is referred to as point-
to-point connection.
• A point-to-point connection provides a dedicated link between the two devices/nodes
as shown in Fig. 1.9. The entire capacity of the link is used for the transmission
between those two devices.
• The point-to-point connection is a unicast connection. There is a dedicated link
between an individual pair of sender and receiver.
• A point-to-point connection is a direct link between two devices/nodes such as a
workstation and a workstation (See Fig. 1.9).
• For example, when we change television channels by using remote control, we
establish a point-to-point connection between the remote control and television.
Link
Fig. 1.9: Point-to-point Connection
2. Multipoint Connection:
• A multipoint connection is a link between more than two devices. It is also known as
multidrop connection.
• In multipoint connection, a single link is shared by multiple devices. So, it can be said
that the channel capacity is shared temporarily by every device connecting to the link.
• The networks having multipoint connections are called broadcast networks. A
broadcast network has a single communication channel that is shared by all the
machines on the network.
• Examples of multipoint connections are Ethernet and Bus topology based on LAN.
1.14
• Fig. 1.10 shows a broadcast network. In Fig. 1.10 we can see that the three
workstations share the common link between the mainframe and the workstations.
Workstation 1 Workstation 2
Link
Mainframe
Workstation 3
Fig. 1.10: Broadcast Connection

• In a multipoint connection environment, the capacity of channel is shared, either
spatially or temporally.
• If several devices can share the link simultaneously, it is called spatially shared
connection. If users must take turns using the link, then it is called temporally shared
or time shared connection.
1.2.4.2 Physical Topology

• The word “topology” comes from “topos”, which is Greek word for “place.” The term
physical topology refers to the way in which a network is laid out physically.
• Network topology refers to the layout of a network and how different nodes in a
network are connected to each other and how they communicate.
• The topology of a network is defined as “the geometric representation of the
relationship of all the links and linking devices (nodes) in a network”.
• A physical topology describes the placement of network nodes and the physical
connections between them.
• The physical topology of a network refers to the configuration of cables, computers,
and other peripherals.
• Bus topology, star topology, ring topology, tree topology, mesh topology, etc. are the
examples of physical topologies.
1.2.4.2.1 Star Topology (April 17, 18, 19, Oct. 18)
• In star topology each device has a dedicated point-to-point link on it to a central

controller, usually called hub or switch. The devices are not directly connected to one
another.
• Each computer on a star network first communicates with a central hub/switch that
forwards the message either to all the computers or only to the destination
computers.
1.15
• Communication is controlled by a central controller (Hub/Switch) only. Star topology

is generally used in LANs.
• Fig. 1.11 shows a star topology or network.
Switch/Hub
Fig. 1.11: Star Topology

Advantages:
1. Easy to install, reconfigure and wire.
2. Centralized management which helps in monitoring the network.
3. Fast as compared to ring topology.
4. Multiple devices can transfer data without collision.
5. Eliminates traffic problems.
6. No disruptions to the network then connecting or removing devices.
7. It is easy to detect the failure and troubleshoot it.
Disadvantages:
1. If the central node (hub or switch) goes down then the entire network goes down.
2. More cabling is required than bus or ring topology, so more expensive.
3. Performance is dependent on capacity of the central device.
1.2.4.2.2 Bus Topology (April 16)
• In bus topology, all nodes are connected to a central cable which is called a bus. This
bus is also called Trunk or sometimes it was also referred to as Backbone cable.
• Trunk cable was then connected to the branch cables which were further connected to
the PCs. Every network device communicates with the other device through this Bus.
• Nodes are connected to the bus cable by drop lines and taps. A drop line is a
connection running between the device and the main cable.
• A tap is a connector that either splices into the main cable or punctures the sheathing
of a cable to create a contact with the metallic core.
• A node (computer) that wants to send data, it puts the data on the bus which carries it
to the destination node.
1.16
• When one computer sends a signal up the wire, all the computers on the network
receive the information, but only one accepts the information.
• The rest regrets the message. One computer can send a message at a time. A computer
must wait until the bus is free before it can transmit.
• Fig. 1.12 shows a bus topology or network.
Laser Printer
Bus/Trunk/Backbone
T-connector
Data flow
Terminator
Terminator
Fig. 1.12: Bus Topology

Advantages:
1. Easy to install and set-up.
2. Requires less cabling length than Mesh and Star so cheaper in cost.
3. Fast as compared to ring topology.
4. Sufficient for a small network.
Disadvantages:
1. It cannot connect a large number of computers.
2. A fault or break in the bus cable stops all transmission.
3. Difficult to identify the problem if the entire network shuts down.
4. Collision may occur.
5. Heavy network traffic can slow a bus considerably.
6. Used for only a small network.
1.2.4.2.3 Ring Topology (April 18, Oct. 17)
• In ring topology, the computers in the network are connected in a circular fashion
which form of a ring.
• In ring topology, each computer is connected to the next computer, with the last one
connected to the first or we can say each device is connected to other two devices with
a dedicated link in one direction, from device to device. Each computer in the ring
incorporates a repeater.
• When a computer receives a signal intended for another computer, its repeater
regenerates the bits and passes them.
1.17
• The message flows around the ring in one direction. Today higher speed LANs has
made this topology less popular.
• Fig. 1.13 shows a ring topology.
Flow of
message
Fig. 1.13: Ring Topology

Advantages:
1. Require less cabling.
2. Less expensive and easy to install.
3. Adding or deleting a device is easy.
4. Reduces chances of collision.
5. Each computer has equal access to resources.
6. Its performance is better than that of Bus topology.
7. Fault isolation is simplified.
Disadvantages:
1. If one node goes down, it takes down the whole network.
2. Slow in speed.
3. Reconfiguration is needed to add nodes, the whole network must be down first.
4. Traffic is unidirectional.
Comparison between Bus, Ring and Star Topologies:
Terms Bus Topology Ring Topology Star Topology
Structure There is a single All computers and There is a central
central cable other devices are host (hub/switch)
(backbone) and all connected in a circle and all nodes
computers and other or ring. connect to it.
devices connect to it.
Contd...
1.18
Host existence Depends on network Depends on network Yes.

needs. needs.
Connection It has no connection Yes. No.
between nodes between the nodes.
Host failure Network can still run. Network will fail. Network will fail.
Trouble-shooting Difficult, there is a Depends on Depends on the
need to search for the backbone. If there is host. It is easier to
problematic node one backbone, trouble- repair the
by one. shooting is difficult. problematic host.
If there is no However, if the
backbone, the focus nodes fail, then
is one the two nodes each node has to be
not communicating. searched.
Ease of adding or Easy. Difficult. Average.
removing nodes
Number of nodes Many. Limited. Limited.
when extending
network
1.2.4.2.4 Mesh Topology (April 16, 17, 18, 19)
• In a mesh network topology, each of the network nodes, computer and other devices,
are interconnected with one another with dedicated point to point link.
• Dedicated means that a link carries traffic only between the two devices it connects. So
for N number of nodes, there will be a total n(n-1)/2 links required.
• Mesh topology is usually implemented in a limited fashion, as a backbone connecting
the main computers of a hybrid network that can include several other topologies.
• Fig. 1.14 shows a mesh topology. Mesh topology is used in WAN.
(a) Full Mesh Topology (b) Partial Mesh Topology
Fig. 1.14: Mesh Topology
1.19
• A mesh topology employs one of two connection arrangements, full mesh topology or
partial mesh topology.
1. In a full mesh network, each network node is connected to every other node in
the network. Due to this arrangement of nodes, it becomes possible for a
simultaneous transmission of signals from one node to several other nodes.
2. In a partially connected mesh network, only some of the network nodes are
connected to more than one node. This is beneficial over a fully connected mesh in
terms of redundancy caused by the point-to-point links between all the nodes.
Advantages:
1. Each connection can carry its own data load due to a dedicated link.
2. Eliminates traffic problems.
3. Mesh topology is robust. If one link becomes unusable, it does not affect other
systems.
4. Privacy or security because of dedicated lines.
5. Point-to-point links make fault identification easy and simple.
Disadvantages:
1. More cables are required than other topologies.
2. Overall cost of this network is very high.
3. Installation and reconfiguration is very difficult.
4. Set-up and maintenance of this topology is very difficult.
5. Expensive due to hardware requirements such as cables.
1.2.4.2.5 Tree Topology

• As its name implies in this topology devices make a tree structure. It is also called a
hierarchical topology.
• Tree topology integrates the characteristics of star and bus topology. In tree topology,
the numbers of star networks are connected using Bus.
• This main cable seems like a main stem of a tree, and other star networks as the
branches.
• Ethernet protocol is commonly used in this type of topology. Fig. 1.15 shows tree
topology.
Advantages:
1. Easy to install and wire.
2. Fast as compared to other topologies.
3. Multiple devices can transfer data without collision.
4. It eliminates traffic problems.
1.20
5. No disruptions to the network then connecting or removing devices.

6. Easy to detect faults and to remove parts.
Disadvantages:
1. It relies heavily on the main bus cable, if it breaks the whole network is crippled.
2. More expensive than other topologies.
3. The cabling cost is more.
4. Scalability of the network depends on the type of cable used.
5. As more and more nodes and segments are added; the maintenance becomes
difficult.
Nodes
Nodes
Concentrator
(Hub)
Backbone cable
Fig. 1.15: Tree Topology
1.2.4.2.6 Hybrid Topology

• Hybrid, as the name suggests, is a mixture of two different things. A hybrid topology is
a combination of two or more network topologies.
• The topology that combines more than one topology is called hybrid topology.
• Two common examples for hybrid networks are star ring network and star bus
network.
1. A star-ring network consists of two or more star topologies connected using a
Multistation Access Unit (MAU) as a centralized hub.
2. A star-bus network consists of two or more star topologies connected using a bus
trunk (the bus trunk serves as the network's backbone).
1.21
• Fig. 1.16 shows an example of star-bus hybrid topology.
Bus
Hub
Hub
Hub
Star
Fig. 1.16: Hybrid Topology
Advantages:
1. Fault detection and troubleshooting is easy in this type of topology.
2. It is easy to increase the size of a network by adding new components, without
disturbing existing architecture.
3. Flexible because this topology can be designed according to the requirements of
the organization and by optimizing the available resources.
4. Hybrid topology is the combination of two or more topologies, so we can design it
in such a way that strengths of constituent topologies are maximized while their
weaknesses are neutralized.
Disadvantages:
1. Hybrid topology is complex and difficult to design.
2. The hubs used to connect two distinct networks are very expensive.
3. As hybrid architectures are usually larger in scale, and so it is time consuming and
difficult to install.
4. Hybrid topology also requires a lot of cables, cooling systems, sophisticated
network devices, etc.
5. Cost is high.
1.2.5 Network Types

• Computer networks fall into three classes regarding the size, distance and the
structure namely LAN (Local Area Network), MAN (Metropolitan Area Network), WAN
(Wide Area Network), as shown in Fig. 1.17.
1.22
Fig. 1.17: Types of Computer Network

• Fig. 1.18 shows geographical arrangement of LAN, WAN and MAN.
WAN
(Overall World
/Globe)
MAN
(Cities, states, etc.)
LAN
(Building,
cyber cafe etc.)
Fig. 1.18: Geographical Arrangement of LAN, WAN and MAN
1.2.5.1 Local Area Network (LAN)

• Local Area Networks (LANs) are privately-owned networks covering a small
geographical area, (less than 1 km) like a home, office or groups of buildings.
• Depending on the needs of the organization and the type of technology used, a LAN
can be as simple as two PCs and a printer or it can extend throughout an organization.
• LANs are widely used to connect personal computers and workstations to share
resources (printers, scanners) and exchange information.
• In LAN computers housed locally within a building or a campus and interlinked by a
single shared medium like cable (See Fig. 1.19).
• LANs are designed to allow resources to be shared between personal computers or
workstations. The resources to be shared can include hardware (e.g., a printer),
software (e.g., an application program) or data.
• LANs are distinguished from other kinds of networks by three characteristics i.e., their
size, their transmission technology and their topology.
1.23
LAN
LAN
Fig. 1.19: A Typical Building LAN
• Generally, LAN will use only one type of transmission medium wired or wireless. The
most common LAN topologies are bus, ring or star.
• Early LAN had data rates in the 4 to 16 mbps range. Today, speeds are normally 100 to
1000 mbps. Wireless LANs are the newest evolution in LAN technology.
• A LAN may be set up using wired or wireless connections. A LAN that is completely
wireless is called Wireless LAN (WLAN).
• WLAN helps us to link single or multiple devices using wireless communication within
a limited area like home, school, or office building. Today most modern day's WLAN
systems are based on IEEE 802.11 standards.
• On most LANs, cables are used to connect nodes like computers, printers, etc. Fig. 1.20
shows a typical LAN.
Fig. 1.20: A Typical LAN
1.24
Advantages of LAN:
1. The reliability of LAN is high because the failure of one computer in the network
does not affect the functioning for other computers.
2. Addition of a new computer to the network is easy and simple.
3. High rate of data transmission is possible.
4. Peripheral devices like magnetic disk, printer etc. can be shared by other
computers.
5. Less expensive to install.
Disadvantages of LAN:
1. Used for small geographical areas (less than 1 km).
2. Limited computers are connected in LAN.
3. Special security measures are needed to stop unauthorized users from using
programs and data.
4. LANs need to be maintained by skilled technicians.
5. In LAN if the file server develops a serious fault, all the users are affected.
1.2.5.2 Metropolitan Area Networks (MAN)

• When the need for connecting a larger number of computers into a single network
arises, the network spreads to a larger area comprising a metropolitan area then it is
known as Metropolitan Area Network (MAN).
• A MAN is a computer network that interconnects users with computer resources in a
geographic region of the size of a metropolitan area like the entire city.
• If a network spans a physical area larger than a LAN but smaller than a WAN, such as
a city then this network is called Metropolitan Area Network (MAN).
• MAN is an extended face of LAN, in which computing devices spread over a city are
interconnected with communication mediums to form a network.
• A MAN connects networks within a city or metropolitan size area. Geographical area
for MAN lies between 5 km to 50 km generally covers towns and cities.
• A MAN is designed for customers who need a high-speed connectivity, normally to the
Internet, and have endpoints spread over a city or part of the city.
• In MAN networks data is transmitted over one or two cables. A good example of a
MAN is the part of the telephone company network that can provide a high-speed DSL
line to the customer.
• The MAN was designed and developed to cover an entire city and the range of MAN is
greater compared to LAN setup. MAN can connect several LANs to form a larger
network (See Fig. 1.21).
1.25
Public city network
Fig. 1.21: MAN

• MAN can be owned by one private organization or public service company such as
local telephone or cable television company.
• By interconnecting smaller networks within a large geographic area, information is
easily disseminated throughout the network.
• Local libraries and government agencies often use a MAN to connect to citizens and
private industries.
• ATM (Asynchronous Transfer Modes) FDDI (Fiber Distributed Data Interface) etc. are
the technologies used in MAN.
Advantages:
1. MAN spans a large geographical area (5 to 50 km) than LAN.
2. MAN falls in between the LAN and WAN therefore, increases the efficiency of
handling data.
3. MAN saves the cost and time attached to establish a wide area network.
4. MAN offers centralized management of data.
5. MAN enables us to connect many fast LANs together.
Disadvantages:
1. Implementation cost is high.
2. Speed is slow.
3. In MAN there are high chances of attacking hackers on the network compared to
LAN. So data may be leaked.
4. To set up MAN it requires technical people that can correctly set up MAN.
1.26
1.2.5.3 Wide Area Networks (WAN)

• The need for the growth of computer networks to global proportions leads to the Wide
Area Network (WAN).
• A large number of autonomous computers are located over a large remote
geographical area called WAN.
• The main feature of WAN is that computers are located in different geographical areas
and each is connected to the main network.
• A WAN provides long distance transmission of data, voice image and
video information over large geographical areas that may comprise a country, a
continent or even the whole world.
• A WAN is a geographically-dispersed collection of LANs. A wide area network is
simply a LAN of LANs or Network of Networks.
• WANs are characterized by the slowest data communication rates and the largest
distances.
• WANs are commonly connected either through the Internet or special arrangements
made with phone companies or other service providers.
• The Internet is the largest WAN, spanning the World today. Fig. 1.22 shows a typical
WAN.
Fig. 1.22: WAN

• WAN contains a collection of machines used for running user (i.e. application)
programs. All the machines called hosts are connected by a communication subnet.
• Fig. 1.23 shows communication Subnet and Host is WAN.
LAN 1
Router LAN 4
LAN 3
LAN 2
Communication subnet
Fig. 1.23: Communication Subnet and Host is WAN

1.27
• The function of the subnet is to carry messages from host to host. The subnet consists
of two important components; namely, transmission lines and switching elements.
• Transmission lines move bits from one machine to another. The switching elements
are specialized computers used to connect two or more transmission lines. When data
arrives on an incoming line, the switching element must choose an outgoing line to
forward them.
• The switching elements are either called as packet switching nodes, intermediate
systems, data switching exchanges or routers.
• When a packet is sent from one router to another via one or more intermediate
routers, the packet is received at an intermediate router. It is stored in the routers
until the required output line is free and then forwarded.
• A subnet using this principle is called a point to point, store-forward or packet
switched subnet.
• WAN's may use public, leased or private communication devices, and can spread over
a wide geographical area. A WAN that is wholly owned and used by a single company
is often called an enterprise network.
• The transmission in WAN is based on a point-to-point technology involving the process
of switching.
Point-to-Point WAN:
• A point-to-point WAN is a network that connects more than two communicating
devices through a transmission media like cable or air.
• The point-to-point WAN is normally a line leased from a telephone or cable TV
provider that connects a home computer or a small LAN to an Internet Service
Provider (ISP).
• A point-to-point WAN is often used to provide Internet access. Fig. 1.24 shows an
example of a point-to-point WAN.
Fig. 1.24: Point-to-Point WAN

Switched WAN:
• The switched WAN connects the end systems and is used in the backbone of global
communication today.
• Switched WAN is a combination of several point-to-point WANs that are connected by
switches.
• Fig. 1.25 shows an example of a switched WAN. Examples of switched WAN includes
asynchronous transfer mode (ATM) network, Wireless WAN.
1.28
Fig. 1.25: Switched WAN
Advantages of WAN:
1. WAN covers a large geographical area.
2. WAN shares software and resources with connecting workstations.
3. Using WAN messages can be sent very quickly to anyone else on the network.
4. Expensive things (like printers or phone lines to the internet etc.) can be shared by
all the computers on the network.
5. WAN adds fluidity to user’s information communication.
Disadvantages of WAN:
1. WANs are expensive.
2. WANs need a good firewall to restrict outsiders from entering and disrupting the
network.
3. Setting up a network can be an expensive and complicated experience. The bigger
the network the more expensive it is.
4. Security is a real issue when many different people have the ability to use
information from other computers. Protection against hackers and viruses adds
more complexity and expense.
5. Slower than LAN and MAN.
Difference between LAN, MAN and WAN: (April 17)
Parameters LAN WAN MAN
1. Stand for Local Area Network. Wide Area Metropolitan Area
Network. Network.
2. Area covered Covers small area Covers large Covers larger areas
i.e. within the geographical than LAN and
building (less than areas, like smaller than WAN
1 km). country, state etc. like cities.
Contd...
1.29
3. Error rates Lowest. Highest. Moderate.

4. Transmission speed High. Low. Moderate .
5. Equipment cost Uses inexpensive Uses most Uses moderately
equipment. expensive expensive
equipment. equipment.
6. Example Offices, Cyber Café. Internet. ATM, FDDI etc.
7. Data transfer rate High. Low. Moderate.
8. Set-up cost Low. High. Moderate.
9. Ownership Owned by private Ownership can be Ownership can be
organizations. private or public. private or public.
10. Designing and Easy and less costly Complex and Complex and more
Maintenance than WAN. more costly than costly than LAN.
LAN.
1.2.5.4 Internetwork (April 16)

• When two or more separate networks are connected for exchanging data or resources,
they become an internetwork (or internet).
• Computer network term is used to describe two or more computers that are linked to
each other. When two or more computer networks or computer network segments are
connected using devices such as a router then it is called computer internetworking.
• Today, it is very rare to see a LAN, a MAN in isolation, they are connected to one
another. When two or more networks are connected, they become an internetwork or
internet.
• An internetwork is formed when distinct networks are interconnected. The internet is
a structured organized system.
• Internetworking started as a way to connect disparate types of computer networking
technology.
• An internetwork is a collection of individual networks, connected by intermediate
networking devices, that functions as a single large network.
• For example, assume that an organization has two offices, one on the east coast and
the other on the west coast.
• Each office has a LAN that allows all employees in the office to communicate with each
other.
• To make the communication between employees at different offices possible, the office
management leases a point-to-point dedicated WAN from a service provider, such as a
telephone company, and connects the two LANs.
1.30
• Now the company has an internetwork, or a private internet (with lowercase i) and
the communication between offices is now possible. Fig. 1.11 shows this internetwork
or internet.
Fig. 1.26: An internetwork made up of Two LANs and One Point-to-Point WAN
• When a host in the west coast office sends a message to another host in the same
office, the router blocks the message, but the switch directs the message to the
destination.
• On the other hand, when a host on the west coast sends a message to a host on the east
coast, router R1 routes the packet to router R2, and the packet reaches the destination.
• Fig. 1.27 shows another internet with several LANs and WANs connected. One of the
WANs is a switched WAN with four switches.
Fig. 1.27: A Heterogeneous Network made of Four WAN and Three LANs
• Today most end users who want Internet connection use the services of Internet
Service Providers (ISPs). There are international, national, regional and local service
providers.
1.31
• There are following two variants of internetwork or internetworking:

1. Intranet: An intranet is a set of interconnected networks or internetworking,
using the Internet Protocol and uses IP-based tools such as web browsers and ftp
tools, that is under the control of a single administrative entity. Intranet is a
private network that is set up within an organization and also controlled by the
organization, nobody outside of the organization is permitted to access the
network.
2. Extranet: An extranet is a network of internetwork or internetworking that is
limited in scope to a single organization or entity but which also has limited
connections to the networks of one or more other usually, but not necessarily,
trusted organizations or entities. Extranet is an extended intranet owned, operated
and controlled by an organization. In addition to allow access to members of an
organization, an extranet uses firewalls, access profiles and privacy protocols to
allow access to users from outside the organization.
Benefits of Internetworking:
1. Internetworks reduces network traffic.
2. The benefit of reduced traffic is optimized performance.
3. Network problems can be more easily identified and isolated in smaller networks,
as opposed to one large network.
4. We can more efficiently span long distances by connecting multiple smaller
networks.
1.2.6 Switching
• An internet is a switched network in which a switch connects at least two links
together. A switch needs to forward data from a network to another network when
required.
• The two most common types of switched networks are circuit-switched network and
packet-switched network.
1.2.6.1 Circuit-Switched Network

• In a circuit-switched network, a dedicated connection, (called a circuit) is always
available between the two end systems; the switch can only make it active or inactive.
• Fig. 1.28 shows a very simple switched network that connects four telephones to each
end.
• We have used telephone sets instead of computers as an end system because circuit
switching was very common in telephone networks in the past, although part of the
telephone network today is a packet-switched network.
1.32
• In Fig. 1.28, the four telephones at each side are connected to a switch. The switch
connects a telephone set at one side to a telephone set at the other side.
• The thick line in Fig. 1.28 connecting two switches is a high-capacity communication
line that can handle four voice communications at the same time; the capacity can be
shared between all pairs of telephone sets.
• The switches used in this example have forwarding tasks but no storing capability.
Fig. 1.28: A Circuit Switched Network

• Let us look at following two cases in a circuit-switched network:
Case 1: In this case, all telephone sets are busy; four people at one site are talking with
four people at the other site; the capacity of the thick line is fully used.
Case 2: In this case, only one telephone set at one side is connected to a telephone set
at the other side; only one-fourth of the capacity of the thick line is used.
• This means that a circuit-switched network is efficient only when it is working at its
full capacity; most of the time, it is inefficient because it is working at partial capacity.
• The reason that we need to make the capacity of the thick line four times the capacity
of each voice line is that we do not want communication to fail when all telephone sets
at one side want to be connected with all telephone sets at the other side.
1.2.6.2 Packet-Switched Network

• In a computer network, the communication between the two ends is done in blocks of
data known as packets.
• In other words, instead of the continuous communication we see between two
telephone sets when they are being used, we see the exchange of individual data
packets between the two computers.
• This allows us to make the switches function for both storing and forwarding because
a packet is an independent entity that can be stored and sent later.
• Fig. 1.29 show a small packet-switched network that connects four computers at one
site to four computers at the other site.
• A router in a packet-switched network has a queue that can store and forward the
packet.
1.33
• Now assume that the capacity of the thick line is only twice the capacity of the data
line connecting the computers to the routers.
• If only two computers (one at each site) need to communicate with each other, there is
no waiting for the packets.
• However, if packets arrive at one router when the thick line is already working at its
full capacity, the packets should be stored and forwarded in the order they arrived.
• The two simple examples show that a packet-switched network is more efficient than a
circuit-switched network, but the packets may encounter some delays.
Fig. 1.29: A Packet-Switched Network
1.2.7 Internet
• The term Internet is derived from the words ‘interconnection’ and ‘networks’. The
Internet is also known as ‘Net’.
• A network is a collection of two or more computers, which are connected together to
share information and resources.
• The Internet is a worldwide system of computer networks, i.e. network of networks.
• Through the Internet, computers become able to exchange information with each
other and find diverse perspectives on issues from a global audience.
• The Internet is the global system of interconnected computer networks that uses
the Internet protocol suite (TCP/IP) to communicate between networks and devices.
• The Internet is a network of networks that consists of private, public, academic,
business, and government networks of local to global scope, linked by a broad array of
electronic, wireless, and optical networking technologies.
• An internet (note the lowercase i) is two or more networks that can communicate with
each other. The most notable internet is called the Internet (uppercase I) and is
composed of thousands of interconnected networks.
• Fig. 1.30 shows a conceptual (not geographical) view of the Internet which shows the
Internet as several backbones, provider networks, and customer networks.
• At the top level, the backbones are large networks owned by some communication
companies like Sprint, Verizon (MCI), AT&T, and so on.
1.34
• The backbone networks are connected through some complex switching systems,
called peering points. At the second level, there are smaller networks, called provider
networks, that use the services of the backbones for a fee.
• The provider networks are connected to backbones and sometimes to other provider
networks. The customer networks are networks at the edge of the Internet that
actually use the services provided by the Internet.
• The customer networks pay fees to provider networks for receiving services.
Backbones and provider networks are also called Internet Service Providers (ISPs).
• The backbones are often referred to as international ISPs, the provider networks are
often referred to as national or regional ISPs.
Fig. 1.30: The Internet Today
1.2.7.1 Accessing the Internet

• Internet access is the ability of individuals and organizations to connect to
the Internet using computer terminals, computers, and other devices; and to access
services such as email and the World Wide Web (WWW).
• Internet access is sold by Internet Service Providers (ISPs) delivering connectivity at a
wide range of data transfer rates via various networking technologies.
• The Internet today is an internetwork that allows any user to become part of it. The
user, however, needs to be physically connected to an ISP. The physical connection is
normally done through a point-to-point WAN.
• In this section we will study various ways to access the Internet.
1.35
1. Using Telephone Networks:

• Today a number of residences and small businesses/organizations have telephone
service, which means they are connected to a telephone network.
• Since, most telephone networks have already connected themselves to the Internet,
one option for residences and small businesses/organizations to connect to the
Internet is to change the voice line between the residence or business and the
telephone center to a point-to-point WAN.
• This can be done in following two ways:
(i) Dial-up Service: Dial-up connection uses telephone lines to connect PCs to the
internet. It requires a modem to set up dial-up connection. This modem works as
an interface between PC and the telephone line. The modem connects the
computer through the standard phone lines, which serves as the data transfer
medium. The modem converts data to voice. The software installed on the
computer dials the ISP and imitates making a telephone connection.
Unfortunately, the dial-up service is very slow, and when the line is used for
Internet connection, it cannot be used for telephone (voice) connection. It is only
useful for small residences. Fig. 1.31 shows typical dial-up connection.
Fig. 1.31: Dial-up Connection/Access of Internet
(ii) DSL Service: Digital Subscriber Line (DSL) service provides a connection to the
Internet through the telephone network. Unlike dial-up, DSL can operate using a
single phone line without preventing normal use of the telephone line for voice
phone calls. DSL uses the high frequencies, while the low (audible) frequencies of
the line are left free for regular telephone communication. Since the advent of the
Internet, some telephone companies have upgraded their telephone lines to
provide higher speed Internet services to residences or small
businesses/organizations. The DSL service also allows the line to be used
simultaneously for voice and data communication. Faster forms of DSL, are also
available like High Data Rate Digital Subscriber Line (HDSL), Very High Data Rate
1.36
Digital Subscriber Line (VHDSL), Asymmetric Digital Subscriber Line (ADSL),

Symmetric DSL (SDSL) and so on. Fig. 1.32 shows a sample connection diagram for
typical residential DSL.
Fig. 1.32: Typical Residential DSL Connection/Access of Internet
Here, are some key points to note in Fig. 1.32:

o Connect your DSL modem's data connection to the phone jack on a wall plate.
o Connect the DSL modem's Ethernet connection to the Ethernet card on your
PC.
o When you connect other telephones or fax machines on the same phone line,
install a micro-filter between the wall plate and each of these devices.
2. Using Cable Networks:
• Now-a-days many cable television companies use some percentage of their network’s
bandwidth to provide internet access through prevailing cable television connections.
• Since this connection uses a special cable modem, it is called "Cable Modem Service". A
cable modem can be added to or integrated with a set-top box that provides the TV set
for Internet access.
• Cable television systems transmit data via coaxial cable, which can transmit data as
much as 100 times faster than common telephone lines.
• More and more residents over the last two decades have begun using cable TV services
instead of antennas to receive TV broadcasting.
• The cable companies have been upgrading their cable networks and connecting to the
Internet. A residence or a small business can be connected to the Internet by using this
service.
• It provides a higher speed connection, but the speed varies depending on the number
of neighbors that use the same cable.
1.37
• Fig. 1.33 shows how internet is accessed using Cable TV connection. A cable modem is
used to access this service, provided by the cable operator.
• The Cable modem comprises two connections: one for internet service and other for
Cable TV signals.
• Since Cable TV internet connections share a set amount of bandwidth with a group of
customers, therefore, data transfer rate also depends on the number of customers
using the internet at the same time.
Fig. 1.33: Typical Residential Cable TV Connection/Access of Internet
3. Using Wireless Networks:

• Now-a-days Wireless connectivity has recently become increasingly popular. A
household or a small business/organization can use a combination of wireless and
wired connections to access the Internet.
• Wireless Internet connection makes use of radio frequency bands to connect to the
Internet and offers a very high speed. The wireless internet connection can be
obtained by either Wi-Fi or Bluetooth.
• Wireless LAN (WLAN) connections are very common these days, which are based on
the technology that is often cited as Wi-Fi (Wireless Fidelity). WLANs are based on
IEEE 802.11 standards.
• To connect to the internet, the wireless access point is connected to a wired LAN like
any other devices, and then computers with wireless NICs can access the wired LAN.
• With the growing wireless WAN access, a household or a small business can be
connected to the Internet through a Wireless WAN (WWAN).
• A WWAN accepts and transmits data using radio signals via cellular sites and
satellites.
• Satellite Internet connection offers high speed connection to the internet. There are
two types of satellite internet connection namely one-way connection or two-way
connection.
• In a one-way connection, we can only download data but if we want to upload, we
need dialup access through an ISP over telephone line.
1.38
Fig. 1.34: WLAN Internet Connection/Access of Internet

• In two-way connection, we can download and upload the data by the satellite. It does
not require any dialup connection.
• Fig. 1.35 shows how the internet is accessed using satellite internet connection.
Fig. 1.35: WWAN Satellite Internet Connection/Access of Internet
4. Direct Connection to the Internet:

• A large organization or a large corporation can itself become a local ISP and be
connected to the Internet.
• This can be done if the organization or the corporation leases a high-speed WAN from
a carrier provider and connects itself to a regional ISP.
• For example, a large university with several campuses can create an internetwork and
then connect the internetwork to the Internet.
1.39
Fig. 1.36: Direct Internet Connection/Access of Internet
• Today, Worldwide Interoperability for Microwave Access (WiMAX) is one of the

popular wireless technology.
Fig. 1.37: WiMax Internet Connection/Access of Internet
1.40
• WiMax is a Wireless MAN (WMAN) technology. WiMAX can provide at-home or

mobile Internet access across whole cities or countries.
• WiMAX would operate similar to Wi-Fi but at higher speed, over greater distances and
for a greater number of users.
• WiMax Internet access is used by businesses/organizations to provide reliable,
dedicated service for Internet access as well as other applications including e-mail, file
sharing, web hosting, data backup, video and so on.
1.2.8 Advantages, Disadvantages and Applications of Network

• The ability to exchange data and communicate efficiently is the main purpose of
networking computers.
Advantages of Computer Network:
1. Easy Communication: A computer network allows all the network users or
computers at a different location to communicate easily. The network users can
communicate with each other using e-mails, instant messaging, video
conferencing, chat rooms, blogging etc.
2. Ability to Share Files, Data and Information: The computer network has the
ability to share data/files and information to users or computers connected to the
computer network.
3. Flexible Access: Access of files from computers throughout the world, and 24 × 7
environment.
4. Backup and Recovery: Generally, in networking the server is placed in a secure
place and the good mechanism is providing for backup of data. If the data is lost
accidentally or due to any other reason, then it is possible to restore them from the
server.
5. Sharing Hardware: By using networking we can share the hardware resources in
an organization and anywhere. For an example, a printer can be shared among
the users in a network so that there’s no need to have individual printers for each
and every computer in the organization. This will significantly reduce the cost of
purchasing hardware.
6. Instant and Multiple Accesses: Computer networking enables multiple users to
access the same data at the same time from a same or remote location. For
example, a World Wide Web (WWW) in which everyone can access a web page
from a different location and read the same information at a same time.
7. Sharing Software: Users can share software within the network easily.
Networkable versions of software are available at considerable savings compared
1.41
to individually licensed versions of the same software. Therefore, large

organizations can reduce the cost of buying software by networking their
computers.
8. Security: Network security issues consist of prevention from virus attacks and
protecting data from unauthorized access. Sensitive files and programs on a
network can be password protected. Only authorized users can access resources in
a computer network.
9. Speed: Sharing and transferring files within networks is very rapid (fast),
depending on the type of network. This will save time while maintaining the
integrity of files.
Disadvantages of Computer Network:
1. Expensive to Build: Building a network is complex and time consuming for large
scale organizations.
2. Virus and Malware: Viruses can spread on a network easily, because of the inter-
connectivity of workstations.
3. Lack of Robustness: If the main file server of a computer network breaks down,
the entire system becomes down and useless.
4. Needs an Efficient Handler: The technical skills and knowledge required to
operate and administer a computer network.
5. High Cost: The investment for hardware and software can be costly for initial set-
up of computer networks.
6. Security Threats: Security threats are always problems with computer networks.
There are hackers who are trying to steal valuable data/information of large
organizations for their own benefit.
Applications of Computer Network:
1. Marketing and Sales: Computer networks are used extensively in both marketing
and sales organizations. Marketing professionals use them to collect, exchange,
and analyze data related to customer needs and product development cycles. Sales
application includes teleshopping, which uses order-entry computers or
telephones connected to order processing networks, and online-reservation
services for hotels, airlines and so on.
2. Financial Services: Today's financial services are totally dependent on computer
networks. Application includes credit history searches, foreign exchange and
investment services, and electronic fund transfer, which allow users to transfer
money without going into a bank (an Automated Teller Machine (ATM) is an
example of electronic fund transfer automatic pay-check is another).
1.42
3. Manufacturing: Computer networks are used in many aspects of manufacturing

including the manufacturing process itself. Two of them that use networks to
provide essential services are Computer-Aided Design (CAD) and Computer-
Assisted Manufacturing (CAM), both of which allow multiple users to work on a
project simultaneously.
4. Directory Services: Directory services allow lists of files to be stored in a central
location to speed worldwide search operations.
5. Information Services: A Network information services includes bulletin boards
and data banks. A World Wide Web (WWW) site offering technical specification
for a new product is an information service.
6. Electronic Data Interchange (EDI): EDI allows business information, including
documents such as purchase orders and invoices, to be transferred without using
paper.
7. Electronic Mail: Probably it's the most widely used computer network application.
8. Teleconferencing: Teleconferencing allows conferences to occur without the
participants being in the same place. Applications include simple text conferencing
(where participants communicate through their normal keyboards and monitor)
and video conferencing where participants can even see as well as talk to other
fellow participants.
9. E-Commerce: Computer networks have paved the way for a variety of business
and commercial transactions online, popularly called e-commerce. Users and
organizations can pool funds, buy or sell items, pay bills, manage bank accounts,
pay taxes, transfer funds and handle investments electronically.
1.3 NETWORK SOFTWARE

• In computer networks, not only the hardware but software is also very important. For
communication between devices, hardware and software both are required.
• Traditional networks were hardware based with software embedded. Network
software is now highly structured.
• Network software encompasses a broad range of software used for design,
implementation, and operation and monitoring of computer networks.
• Networking software, in the most basic sense, is software that facilitates, enhances or
interacts with a computer network.
• One type of networking software allows computers to communicate with one another,
while another type of networking software provides users access to shared programs.
• Networking software is a key component of today's computer networks, including the
Internet.
1.43
• Network software is a general phrase for software that is designed to help set up,
manage, and/or monitor computer networks.
• Networking software applications are available to manage and monitor networks of
all sizes, from the smallest home networks to the largest enterprise networks.
Functions of Network Software:
1. Helps to set up and install computer networks.
2. Enables users to have access to network resources in a seamless manner.
3. Allows administrations to add or remove users from the network.
4. Helps to define locations of data storage and allows users to access that data.
5. Helps administrators and security systems to protect the network from data
breaches, unauthorized access and attacks on a network.
• With the advent of Software Defined Networking (SDN), software is separated from
the hardware thus making it more adaptable to the ever-changing nature of the
computer network.
1.3.1 Protocol Hierarchies

• In a computer network, many devices are connected to each other. Every computer is
working as a source or destination or intermediate machine.
• It is also possible that a computer is simultaneously performing all these jobs. As a
source, destination or intermediate machine certain predefined jobs a computer has to
perform. The entire structure of a computer network is complex.
• To reduce their design complexity, networks are organized as a stack of layers or
levels, each one built upon the one below it.
• The number of layers, the name of each layer, the contents of each layer and the
function of each layer differ from network to network. Each layer offers certain
predefined services to higher layers and taking services from lower layers.
• The basic concept of layering network responsibilities is that each layer adds value to
services provided by sets of lower layers.
• In this way, the highest level is offered the full set of services needed to run a
distributed data application.
• Fig. 1.38 shows a layered architecture where layer n at the source logically (but not
necessarily physically) communicates with layer n at the destination and layer n of
any intermediate nodes.
• Layer n on one machine carries on a conversation with layer n on another machine.
The rules and conventions used in this conversation are collectively known as the
layer n protocol.
1.44
• A protocol is an agreement between the communicating parties on how

communication is to proceed.
• A five-layer network is shown in Fig. 1.38. The entities comprising the corresponding
layers on different machines are called peers.
• The peers may be processes, hardware devices or even human beings. It is the peers
that communicate by using the protocol.
• In reality, no data are directly transferred from layer n on one machine to layer n on
another machine.
• Each layer passes data and control information to the layer immediately below it, until
the lowest layer is reached. Below layer 1 is the physical medium through which
actual communication occurs.
• In Fig. 1.38 virtual communication is shown by dotted lines and physical
communication by solid lines.
• Between each pair of adjacent layers is an interface. The interface defines which
primitive operations and services the lower layer gives to the upper layer.
• A set of layers and protocols is called a network architecture.
• The details of implementation and the specification of the interfaces is not part of
architecture because they are hidden inside the machines and not visible from
outside.
• A list of protocols used by a certain system, one protocol per layer, is called a protocol
stack.
Fig. 1.38: Layers, Protocols and Interfaces

1.45
• To understand the idea of multilayer communication, consider the example of post

office.
Fig. 1.39
• Now, we will discuss a more technical example of how to provide communication to

the top layer of the five-layer network.
• A message M, is produced by an application process running in layer 5 and given to
layer 4 for transmission. Layer 4 adds a header part i.e. control information in front of
the message and passes the message to layer 3.
• The control information may include sequence number, source address, destination
address, total number of bytes etc.
• We assume that layer 3 is not able to handle large data, so layer 3 breaks-up the
incoming messages into smaller units (Packets), adding layer 3 header to it and gives
to layer 2. In our example, M is split into two parts M1 and M2.
• Layer 2 adds not only a header, but also a trailer (which is also a control information)
to it and passes to layer no. 1 for physical transmission.
• At the receiving machine, the message is received by layer 1 and then moves upward
from layer to layer.
1.46
• Every layer removes the header part attached by the corresponding layer from the
sender's machine. None of the headers for layers below n are passed up to layer n.
Layer
Layer 5 protocol
M M
5
Layer 4 protocol
H4 M H4 M
4
Layer 3 protocol
3 H 3 H 4 M1 H 3 M2 H 3 H 4 M1 H 3 M2
Layer 2
protocol
2 H2 H3 H4 M1 T2 H2 H3 M2 T2 H2 H3 H4 M1 T2 H2 H3 M2 T2
1
Source machine Destination machine
Fig. 1.40: Data Flow between Machines
1.3.2 Design Issues of the Layers

• A network consists of a series of levels called layers. The various key design issues are
present in several layers in computer networks.
• Some of the main design issues in layers are as follows:
1. Addressing: Every layer needs a mechanism for identifying senders and receivers.
In computer networks from many computers, some of which have multiple
processes, some sort of mechanism is needed for a process on one machine to
specify with whom it wants to talk. As a consequence of having multiple
destinations, some form of addressing is needed in order to specify a specific
destination.
2. Data Transfer Methods: Data transfer means sending data from one computer to
another. In computer networks, data transfer can be of three modes i.e. simplex,
half duplex and full duplex.
3. Error Control: Error control is an important issue because physical
communication channels are not perfect. Many error-detecting and error-
correcting codes are known, but both ends of the connection must agree on which
one is being used. The receiver must have some way of telling the sender which
messages have been correctly received and which have not.
4. Flow Control: If the sender is fast compared to the receiver, the issue occurs at
every layer is how to keep a fast sender from swamping a slow receiver with data.
1.47
One solution is both entities agree upon the data rate to control the flow. Another
solution involves some kind of feedback from the receiver to the sender about the
receiver’s current situation. This subject is called flow control.
5. Disassembling and Reassembling of Messages: Another problem that must be
solved at several levels is the inability of all processes to accept arbitrarily long
messages. One solution is to divide long messages into multiple small messages,
transmit them and combine them at the receiving end.
6. Multiplexing and Demultiplexing: When it is inconvenient or expensive to set up
a separate connection for each pair of communicating processes, the underlying
layer may decide to use the same connection for multiple, unrelated
conversations. This is known as multiplexing. Multiplexing is needed in the
physical layer.
7. Routing: When multiple paths between source and destination are available, a
route must be chosen depending upon certain criteria. Criteria may be current
traffic or situation of the subnet. Criteria is called a routing protocol and this
process is called routing.
8. Order of Messages: Not all communication channels preserve the order of
messages sent on them. To deal with possible loss of sequencing, the protocol must
make explicit provision for the receiver to allow the pieces to be reassembled
properly.
9. Scalability: Networks are continuously evolving. The sizes are continually
increasing leading to congestion. Also, when new technologies are applied to the
added components, it may lead to incompatibility issues. Hence, the design should
be done so that the networks are scalable and can accommodate such additions
and alterations.
10. Security: A major factor of data communication is to defend it against threats like
eavesdropping and surreptitious alteration of messages. So, there should be
adequate mechanisms to prevent unauthorized access to data through
authentication and cryptography.
11. Resource Allocation: Computer networks provide services in the form of network
resources to the end users. The main design issue is to allocate and deallocate
resources to processes. The allocation/deallocation should occur so that minimal
interference among the hosts occurs and there is optimal usage of the resources.
1.3.3 Connection-Oriented and Connection-less Services

• Layers can offer two types of services to the layers above them namely connection-
oriented service and connection-less service.
1.48
1.3.3.1 Connection-Oriented Services

• In connection-oriented service, logical connection is established between
communicating parties. Connection oriented service is modeled after the telephone
system.
• In the telephone system, when we want to make a call, we have to pick up the phone,
dial the number, after that connection is established, use that connection and then
disconnect the telephone.
• To use a connection oriented network service, the service user first establishes a
connection, uses the connection and then releases the connection. Once, the
connection is established between source and destination, the path is fixed.
• The data transmission takes place through this path established. The order of the
messages sent will be the same at the receiver end.
• Services are reliable and there is no loss of data. Most of the time, reliable service
provides acknowledgement as an overhead and adds delay.
Basic Working Concept of Connection-Oriented Service:
• The connection-oriented service first establishes the virtual connection between the
source and the destination.
• Then transfers all data packets belonging to the same message through the same
dedicated established connection and after all packets of a message are transferred it
releases the connection.
Fig. 1.41: Connection-oriented Service
• To establish a connection a source sends a request packet to the destination. In

response to which destination sends the acknowledgement packet to the source
confirming that the destination is ready to accept the data from the source.
1.49
• The routers involved in the exchange of request and acknowledgement packets

between source and destination, define the virtual path that will be followed by all
packets belonging to the same message.
Advantages of Connection-oriented Services:
1. These services provide guaranteed delivery of data.
2. This service is more reliable than connectionless services.
3. Some connection oriented services will monitor for lost packets and handle
resending them.
Disadvantages of Connection-oriented Services:
1. A connection must require.
2. These services have more overhead than connectionless service.
3. Complex method for data transferring.
1.3.3.2 Connection-less Services

• Connection-less service is modeled after the postal system. In this type of services, no
connection is established between source and destination.
• Here, there is no fixed path. Therefore, the messages must carry a full destination
address and each one of these messages are sent independent of each other.
• Messages sent will not be delivered at the destination in the same order. Thus,
grouping and ordering is required at the receiver end, and the services are not
reliable.
• There is no acknowledgement confirmation from the receiver. Unreliable
connectionless service is often called datagram service, which does not return an
acknowledgement to the sender.
• In some cases, establishing a connection to send one short message is needed. But
reliability is required, and then acknowledgement datagram service can be used for
these applications.
Basic Working Concept of Connection-less Service:
• Connection-less service is a method of data transmission between two computers in a
different network. Connectionless service is also termed as datagram service.
• The connection-less service look-alike the postal system where each letter carries its
source and destination address and each one of them is routed through a different
path.
• The source divides the message into small acceptable packets known as a datagram.
These datagrams are individually pushed into the network; each datagram may travel
a different path.
1.50
• The network considers each datagram or data packet as an independent entity i.e., no
relationship is considered between the packets belonging to the same message.
• Each datagram carries its source and destination address. The router uses the
destination address to route the datagram to its destination.
• The packets received at the destination may be received out of order. Hence, the
datagrams are assembled to recreate the original message.
Fig. 1.42: Connection-less Service

Advantages of Connectionless Services:
1. Does not requires any connection.
2. These services are very simple and easy for data transfer.
3. Used for periodic burst data transfer.
4. Less overhead than connection oriented services.
Disadvantages of Connectionless Services:
1. Less reliable than connection-oriented services.
2. No guarantee for delivery of data.
3. It provides minimal services.
• Following table describes various services and examples of connection oriented and
connectionless services.
Service Example
Connection- Reliable message stream Sequence of pages
Oriented Reliable byte stream Remote login
Unreliable connection Digitized voice
Unreliable datagram Electronic junk mail
Connection- Acknowledged datagram Registered mail
Less Request-reply Database query
1.51
Difference between Connectionless and Connection-oriented Services:

Sr. Connectionless Connection-oriented
Characteristic
No. Service Service
1. Connection setup Data is sent without setup i.e. Connection must be

connectionless. established.
2. Example of UDP (User Datagram Protocol) TCP (Transmission Control
protocol Protocol).
3. General Simple, high-speed, low Full-featured protocol that
description functionality "wrapper" that allows applications to send
interfaces applications to the data reliably without
network layer. worrying about network
layer issues.
4. Data interface to Message-based; data is sent in Stream-based; data is sent
application discrete packages by the by the application with no
applications. particular structure.
5. Reliability and Unreliable, best efforts Reliable delivery of
acknowledgments delivery without messages; all data is
acknowledgments acknowledged.
6. Retransmission Not performed. Application Delivery of all data is
must detect lost data and managed and lost data is
retransmit if needed. retransmitted
automatically.
7. Features provide None. Flow control using sliding
to manage flow of windows; window size
data. adjustment heuristics
congestion avoidance
algorithm.
8. Data delivery Connectionless services do Connection-oriented
guarantee not. services provide some level
of delivery guarantee.
9. Overhead Connectionless networks have Connection-oriented
less overhead. network services have
more overhead.
10. Transmission Very high. Low.
speed
Contd....
1.52
11. Data quantity Small to moderate amounts of Small to very large

suitability data. amounts of data.
12. Packet Route Packets can follow any route. Packets follow the same
route.
1.4 REFERENCE MODELS

• A reference model in networking is a conceptual layout that describes how
communication between devices should occur.
• In order to provide communication among heterogeneous devices, we need a
standardized model i.e. a reference model, which would provide us how these devices
can communicate regardless of their architecture.
• The purpose of the reference/network model was to define an architectural
framework that defines the logical communication tasks that are required to move
information between different computer systems.
• For efficient communication, the reference model identifies the tasks involved in
inter-computer communication and divides them in logical groups called layers, with
each layer performing a specific function.
• A communication system designed in such a manner is referred to as layered
architecture. We used the concept of layers in our daily life, for example, consider
communication between two friends through postal mail.
Sender Receiver
The letter is written, The letter is picked up,

put in an envelope, and Higher layers removed from the
dropped in a mailbox envelope, and read
The letter is carried The letter is carried

from the mailbox Middle layers from the post office
to a post office to a mailbox
The letter is delivered The letter is delivered

to a carrier by the post Lower layers form the carrier
office to the post office
Fig. 1.43: Communication Process
1.53
• The process of sending a letter by one friend to another is difficult, if there are no
services available by post office.
• In the Fig. 1.43, we have a sender, a receiver and a carrier. At sender and receiver’s
end, all the activities done are grouped in three layers.
• The task of transporting the letter between the sender and receiver is done by the
carrier. At the sender site, the letter must be written and dropped in the mailbox
before being picked up by the letter carrier and delivered to the post office.
• At the receiver site, the letter must be dropped in the recipient mailbox before being
picked up and read by the recipient. Every step must be carried out sequentially one
by one. In the same way computers also work. Each layer at sending and site uses
services of the layer immediately below it.
• There are two computer network models namely, OSI Model and TCP/IP Model on
which the whole data communication process relies. In this section, we will study
these two network models with layers and functions of each layer.
1.4.1 OSI Reference Model (Oct. 17)

• In 1978, the International Organization for Standardization (ISO) developed an
architecture that would allow the devices of different manufacturers to work together
to communicate with different operating systems.
• In 1984, the ISO architecture became an international standard known as the Open
Systems Interconnection (OSI) reference/network model.
• The model is called the ISO-OSI reference model because it deals with connecting open
systems i.e. systems that are open for communication with other systems.
• The purpose of the OSI model is to show how to facilitate communication between
different systems without requiring changes to the logic of the underlying hardware
and software.
• The OSI model is not a protocol, it is a model for understanding and designing a
network architecture that is flexible, robust and interoperable.
• The OSI model is a layered framework for the design of network systems that allows
communication between all types of computer systems.
• The OSI model consists of seven separate but related layers, each of which defines a
part of the process of moving information across a network.
• The seven layers of OSI model are physical layer, data link layer, network layer,
transport layer, session layer, presentation layer and application layer.
• Fig. 1.44 shows OSI reference/network model. The principles that were applied to
arrive at the seven layers are:
1. A layer should be created where a different level of abstraction is needed.
2. Each layer should perform a well-defined function.
1.54
3. The function of each layer must support internationally standardized protocols.

4. The layer boundaries should be chosen to minimize the information flow across
the interfaces.
7 Application
6 Presentation
5 Session
4 Transport
3 Network
2 Data Link
1 Physical
Fig. 1.44: OSI Reference Model
• The OSI model allows complete interoperability between incompatible systems.

Within a single machine, each layer calls upon the services of the layer just below it.
• For example, layer 4 uses the services of layer 3 and gives services to layer 5. Between
machines, layer X on one machine communicates with layer X on another machine.
• This communication is governed by an agreed-upon series of rules and conventions
called protocols. The processes on each machine that communicates at a given layer
are called peer-to-peer processes.
• Layers of the OSI model are divided into two groups namely, upper layer and lower
layer.
• The upper OSI layers are almost always implemented by software while lower layers
are a combination of hardware and software, except the physical layer, which is
mostly hardware.
• The upper layers of the OSI model deal with application issues and generally are
implemented only in software.
• Generally speaking, software in these layers performs application-specific functions
like data formatting, encryption, and connection management.
• The lower layers of the OSI model provide more primitive network-specific functions
like routing, addressing, and flow control.
• Upper layers in OSI model are:
7. Application layer,
6. Presentation layer, and
5. Session layer.
1.55
• Lower layers in OSI model are:

4. Transport layer,
3. Network layer,
2. Data link layer and
1. Physical layer.
• The main benefits of the OSI model include the following:
1. Helps users understand the big picture of networking.
2. Makes troubleshooting easier by separating networks into manageable pieces.
3. Defines terms that networking professionals can use to compare basic functional
relationships on different networks.
4. Helps users understand how hardware and software elements function together.
5. Helps users understand new technologies as they are developed.
1.4.1.1 Basic Concepts of OSI Model

• In this section we will study basic concepts in the OSI reference model.
1. Peer-to-Peer Processes:
• The entities comprising the corresponding layers on different computer machines are
called peers.
• Within each machine, a layer calls upon the services of the layer below it while
providing its own services to the layer above.
Intermediate Intermediate
Device X node node
Device Y
Peer-to-peer protocol (7th layer)

7 Application Application 7
7-6 interface 7-6 interface
Peer-to-peer protocol (6th layer)
6 Presentation Presentation 6
6-5 interface Peer-to-peer protocol (5th layer) 6-5 interface
5 Session Session 5
5-4 interface Peer-to-peer protocol (4th layer) 5-4 interface
4 Transport Transport 4
4-3 interface 3rd 3rd 3rd 4-3 interface
3 Network Network Network Network 3
3-2 interface 2nd 2nd 2nd 3-2 interface
2 Data link Data link Data link Data link 2
2-1 interface 1st 1st 1st 2-1 interface
1 Physical Physical Physical Physical 1
Physical communication
Fig. 1.45: Interactions between OSI model layer or Layered Architecture of OSI Model
1.56
• At the physical layer, communication is direct i.e., Machine X sends a stream of bits to
machine Y.
• At the higher layers, however, communication must move down through the layers on
machine X, over to machine Y, and then back up through the layers.
• Each layer in the sending machine adds its own information to the message it receives
from the layer just above it and passes the whole package to the layer just below it.
This information is added in the form of headers or trailers (control data appended to
the beginning or end of a data parcel).
• Headers are added to the message at layers 6, 5, 4, 3, and 2. X trailer is added at
layer 2.
• Headers are added to the data at layers 6, 5, 4, 3, and 2. Trailers are usually added only
at layer 2. At layer 1 the entire package is converted to a form that can be transferred
to the receiving machine.
• At the receiving machine, the message is unwrapped layer by layer, with each process
receiving and removing the data meant for it.
• For example, layer 2 removes the data meant for it, then passes the rest to layer 3.
Layer 3 removes the data meant for it and passes the rest to layer 4, and so on.
Sender Receiver
L7 data 7 7 L7 data
L7 data H6 6 6 L7 data H6
T2 T2
1 1
0 0101010101101010000010000 0 0101010101101010000010000
Transmission medium
Fig. 1.46: An exchange using OSI model
1.57
2. Interfaces between Layers:

• The passing of the data and network information down through the layers of the
sending machine and back up through the layers of the receiving machine is made
possible by an interface between each pair of adjacent layers.
• Each interface defines what information and services a layer must provide for the
layer above it.
• Well-defined interfaces and layer functions provide modularity to a network.
3. Organization of the Layers:
• All the seven layers are grouped into three subgroups. Layer 1, 2 and 3 are called
network support layers and deal with the physical aspect of moving data from any
device to another.
• Layer 5, 6 and 7 are called user support layers, and allow interoperability among
unrelated software systems.
• Layer 4 links the two subgroups and ensures that what lower layers have transmitted
is in a form that the upper layers can use.
1.4.1.2 Functions of Each Layer of OSI Model

• In this section, we will discuss the functions of each layer in the OSI model.
1. Physical Layer: (April 18)
st
• The physical layer is the lowest layer (1 ) of the OSI model. The physical layer
coordinates the functions required to carry a bit stream over a physical medium.
• Physical layer deals with the mechanical and electrical specifications of the interface
and transmission medium.
• The physical layer also concerned with the following functions:
(i) Physical Characteristics of Interfaces and Medium: Physical layer defines the
characteristics of the interface between the devices and the transmission
medium. It also defines the type of transmission medium.
(ii) Representation of Bits (Data Encoding): Below the physical layer there is a
transmission medium, which carries computer data. Any transmission medium
does not understand about computer data i.e. 0 and 1, it understands only about
signals. Physical layer converts binary data into signals and vice versa. For this
different types of encoding methods are used by the Physical layer.
(iii) Data Rate Control: Physical layer defines the transmission rate i.e. the number of
bits sent in one second. Therefore, it defines the duration of a bit.
(iv) Bit synchronization: The physical layer provides the synchronization of the bits
by providing a clock. This clock controls both transmitter as well as receiver thus
providing synchronization at bit level.
1.58
(v) Line Configuration: Physical layer also defines the way in which the devices are
connected to the medium. Two different line configurations are used point to
point configuration and multipoint configuration.
(vi) Physical Topology: The physical topology defines how devices are connected to
form a network. Devices can be connected by using star, ring, mesh, bus etc.
topologies.
(vii)Transmission Mode: Physical layer also defines the way in which the data flows
between the two connected devices. The various transmission modes possible are
simplex, half-duplex and full-duplex.
• The major protocols used by the physical layer include Bluetooth, OTN (Optical
Transport Network), DSL, IEEE.802.11, IEEE.802.3, and so on. Hub, Repeater, Modem,
Cables etc., are physical layer connecting devices.
2. Data Link Layer:
nd
• The 2 layer of the OSI model is the data link layer. The goal of the data link layer is to
provide reliable, efficient communication between adjacent machines connected by a
single communication medium.
• Data link layer sends data frames from the Network layer to the Physical layer. The
data link layer divides the stream of bits received from the network layer to
manageable data units called frames.
• Data link layer is divided into following two sub layers:
(i) Media Access Control (MAC): The MAC sub layer controls the means by which
multiple devices share the same media channel. This includes contention
methods and other media access details. The MAC layer also provides addressing
information for communication between network devices.
(ii) Logical Link Control (LLC): The LLC sub layer establishes and maintains links
between communicating devices. LLC sublayer provides interface between the
media access methods.
• Functions of data link layer are given below:
(i) Framing: Framing provides a way for a sender to transmit a set of bits that are
meaningful to the receiver. Data link layer performs various framing functions
like frame traffic control, frame sequencing, frame delimiting, frame error
checking and so on.
(ii) Physical Addressing: After creating frames, the data link layer adds physical
addresses (MAC address) of sender and/or receiver in the header of each frame.
(iii) Flow Control: Flow control is the traffic regulatory mechanism implemented by a
data link layer that prevents the fast sender from drowning the slow receiver.
1.59
(iv) Error Control: It provides the mechanism of error control in which it detects and
retransmits damaged or lost frames. Data link layer also deals with the problem
of duplicate frames, thus providing reliability to the physical layer.
(v) Media Access Management: Data link layer determines when the node "has the
right" to use the physical medium.
(vi) Access Control: When two or more devices are connected to the same link, data
link layer protocols decide which device has control over the link at a given time.
Means data link layer protocols decides which device is going to use the link (for
transmission etc.) at what time.
• The protocols are used by the Data Link Layer includes Point-to-Point Protocol (PPP),
Point-to-Point High-Level Data Link Control (HDLC), Serial Line Internet Protocol
(SLIP), Spanning Tree Protocol (STP,) Address Resolution Protocol (ARP), IEEE.802.3,
ARCNET etc. Switch and Bridge are data link layer connecting devices.
3. Network Layer:
rd
• The 3 layer of the OSI model is the network layer. The network layer is responsible
for the delivery of individual packets from the source host to the destination host.
• When source and destination are from the same network, there is usually no need for
network layer, delivery of packets is handled by data link layer.
• However, if source and destination are from different network, network layer is
responsible for delivery of data packets.
• Functions of network layer are given below:
(i) Logical Addressing: Large numbers of different networks can be combined
together to form bigger networks or internetwork. In order to identify each
device on internetwork uniquely, the network layer defines a logical addressing
scheme, which distinguishes each device uniquely and universally on the
Internet. When source and destination are from different networks, to deliver the
packet logical addresses (IP) are required.
(ii) Routing: When independent networks or links are combined together to create
internet works, multiple routes are possible from source machine to destination
machine. The network layer protocols determine which route or path is best from
source to destination. This function of the network layer is known as routing.
Routes frames among networks.
(iii) Congestion Control: This layer is also responsible for handling the congestion
problem at the node, when there are too many packets stored at the node to be
forwarded to the next node.
(iv) Internetworking: One of the main responsibilities of a network layer is to
provide internetworking between different networks. It provides a logical
connection between different types of network.
1.60
(v) Packetizing: The network layer receives the data from the upper layers and
creates its own packets by encapsulating these packets. The process is known as
packetizing. This packetizing is done by Internet Protocol (IP) that defines its own
packet format.
(vi) Fragmentation: Fragmentation means dividing the larger packets into small
fragments. The maximum size for a transportable packet is defined by physical
layer protocol. For this, network layer divides the large packets into fragments so
that they can be easily sent on the physical medium.
• The network layer uses protocols such as Internet Protocol (IP), Internet Protocol
version 4 (IPv4), Internet Protocol version 6 (IPv6), Internet Control Message
Protocol (ICMP), Internet Group Management Protocol (IGMP), Internet Protocol
Security (IPsec), Internetwork Packet Exchange (IPX), Routing Information
Protocol (RIP) etc. Network layer uses networking connecting devices such as routers
and gateway.
4. Transport Layer:
th
• The 4 layer of the OSI model is the transport layer. Transport layer is responsible for
the process to process delivery of the entire message. A process is an application
program running on a host.
• Transport layer ensures that packets are delivered error free, in sequence with no
losses or duplications. Transport layer unpacks, reassembles and sends receipt of
messages at the receiving end.
• Transport layer provides flow control, error handling, and solves transmission
problems. The transport layer is responsible for source-to-destination, (end-to-end)
delivery of the entire message.
• Transport layer provides following two types of services:
(i) In Connection Oriented Transmission the receiving device sends an
acknowledgment, back to the source after a packet or group of packet is received.
This type of transmission is also known as a reliable transport method. Because
connection oriented transmission requires more packets to be sent across the
network, it is considered a slower transmission method.
(ii) In Connectionless Transmission the receiver does not acknowledge receipt of a
packet that the packet arrives just fine. This approach allows for much faster
communication between devices.
• Functions of transport layer are given below:
(i) Service Point Addressing (Port Addressing): The purpose of the transport layer
is to deliver messages from one process running on source machine to another
process running on destination machine. It may be possible that several
1.61
programs or processes are running on both the machines at a time. In order to

deliver the message to the correct process, the transport layer header includes a
type of address called service point address or port address. Thus by specifying
this address, the transport layer makes sure that the message is delivered to the
correct process on the destination machine.
(ii) Segmentation and Reassembly: Transport layer accepts a message from the
(session) layer above it, splits the message into smaller units (if not already small
enough), and passes the smaller units down to the network layer. The transport
layer at the destination station reassembles the message.
(iii) Connection Control: The transport layer can provide connection oriented or
connectionless services for connection control. A connectionless transport layer
treats each segment as an independent packet and delivers it to the transport
layer at the destination. A connection oriented transport layer makes a
connection with the destination transport layer first and then delivers data. After
all data transfer is done the connection is terminated.
(iv) Flow Control: Like Data link layer, transport layer also performs flow control.
Transport layer makes sure that the sender and receiver communicate at a rate
they both can handle. Therefore, flow control prevents the source from sending
data packets faster than the destination can handle. Flow control performed by
the transport layer is end to end.
(v) Error Control: Like Data link layer, Transport layer also performs error control.
Here error control is performed end-to-end rather than across a single link. Error
correction is usually achieved through retransmission.
5. Session Layer: (April 17)
th
• The 5 layer of the OSI model is the session layer. Session layer has the primary
responsibility of beginning, maintaining and ending the communication between two
devices, which is called Session.
• Session layer also provides for orderly communication between devices by regulating
the flow of data.
• The session layer is the network dialog controller. It establishes, maintains and
synchronizes the interaction among communicating systems.
• Function of session layer are given below:
(i) Dialog Control: Dialog control is the function of the session layer that determines
which device will communicate first and the amount of data that will be sent. It
also decides the communication between two processes to take place in either
half duplex of full duplex mode.
1.62
(ii) Dialog Separation or Synchronization: Session layer allows a process to add

synchronization points or check points, to a stream of data. The session layer
decides the order in which data need to be passed to the transport layer.
(iii) Session Establishment, Maintenance and Termination: It allows two
application processes on different machines to establish, use and terminate a
connection, called a session.
(iv) Session Support: It performs the functions that allow these processes to
communicate over the network, performing security, name recognition, logging
and so on.
• The protocols used in session layer includes Layer 2 Tunneling Protocol (L2TP), Point-
to-Point Tunneling Protocol (PPTP), Network Basic Input Output System (NetBIOS),
PAP (Password Authentication Protocol), Sockets Direct Protocol (SDP) etc.
6. Presentation Layer: (April 17)
th
• The 6 layer of the OSI model is the presentation layer. The Presentation layer is also
called Translation layer.
• The presentation layer presents the data into a uniform format and masks the
difference of data format between two dissimilar systems.
• The presentation layer is concerned with the syntax and semantics of the information
transmitted between two systems. Presentation layer is responsible for translation,
compression and encryption.
• Functions of presentation layer are given below:
(i) Translation: The translation between the sender and the receiver’s message
formats done by the presentation layer if the two formats are different.
(ii) Encryption: Converting computer data into non-readable form is encryption. It is
required for important data transmission. Decryption reverses the original
process to transform the message back to its original form.
(iii) Compression: Data compression reduces the number of bits contained in the
information. Data compression becomes particularly important in the
transmission of multimedia such as text, audio and video.
• The presentation layer uses protocols Multipurpose Internet Mail Extensions (MIME)
Network News Transfer Protocol (NNTP), Transport Layer Security (TLS) and Secure
Sockets Layer (SSL) etc.
7. Application Layer:
th
• The 7 layer of the OSI model is the application layer. The application layer enables
the user (human or software), to access the network.
1.63
• Application layer provides user interfaces and support for services such as e-mail,
remote file access and transfer, shared database management and other types of
distributed information services.
• Functions of application layer are given below:
(i) Network Virtual Terminal: A network virtual terminal is a software version of a
physical terminal and it allows a user to log onto a remote host. The remote host
believes it is communicating with one of its own terminals and allows the user to
log on.
(ii) File Transfer, Access and Management (FTAM): This application allows a user
to access files in a remote host, to retrieve files from a remote host for use in the
local computer and to manage or control files in a remote computer locally.
(iii) Mail Services: This application provides e-mail operations like forwarding and
storage.
(iv) Remote Logins: This layer allows logging into a host which is remote.
(v) Directory Services: This application provides distributed database sources and
access for global information about various objects and services.
• Application layer uses protocols such as Secure Shell (SSH), File Transfer
Protocol (FTP), Trivial File Transfer Protocol (TFTP), Simple Mail Transfer
Protocol (SMTP), Internet Message Access Protocol (IMAP), Reverse Address Resolution
Protocol (RARP), Bootstrap Protocol (BOOTP), Simple Network Management
Protocol (SNMP), Border Gateway Protocol (BGP), Hypertext Transfer Protocol (HTTP)
etc.
1.4.2 TCP/IP Reference Model

• TCP/IP stands for Transmission Control Protocol/Internet Protocol. The TCP/IP model
was developed by the U.S. Department of Defense (DoD) to connect multiple networks
and preserve data integrity.
• The TCP/IP is the conceptual model and set of communications protocols used on the
Internet and similar computer networks.
• The TCP/IP protocol suite establishes the technical foundation of the Internet. TCP/IP
protocol suite also called the Internet protocol suite.
• The TCP/IP is a protocol suite (a set of protocols organized in different layers) used in
the Internet today. It is a hierarchical protocol made up of interactive modules, each of
which provides a specific functionality.
• The term hierarchical means that each upper level protocol is supported by the
services provided by one or more lower level protocols.
1.64
• The original TCP/IP protocol suite was defined as four software layers built upon the
hardware. Today, however, TCP/IP is thought of as a five-layer model as shown in
Fig. 1.47.
Fig. 1.47: Layers in the TCP/IP Protocol Suite
• Fig. 1.48 shows five-layer TCP/IP reference model.
Fig. 1.48: Five Layers in the TCP/IP Reference Model
1.4.2.1 Layered Structure of TCP/IP

• TCP/IPs an industry-standard protocol suite for Wide Area Networks (WANs)
developed in the 1970s and 1980s by the U.S. Department of Defense (DoD).
• TCP/IP is not one protocol, but is a suite of many protocols. The protocols define
applications, transport controls, networking, routing, and network management.
• The TCP/IP is a set of protocols, or a protocol suite, that defines how all transmissions
are exchanged across the Internet.
• TCP/IP protocol suite contains five layers. To better understand the duties of each
layer, we need to think about the logical connections between layers.
• Fig. 1.49 shows logical connections in simple internet.
1.65
Source host Destination host

Logical connections
Application Application
Transport Transport
Network Network
Data link Data link
Physical Physical
Switch Router Switch
LAN LAN
Router
Source Link 1 Link 2 Destination

host To link 3 host
Fig. 1.49: Logical Connection between Layers of the TCP/IP Protocol Suite
• Using logical connections in TCP/IP protocol suite makes it easier for us to think about
the duty of each layer.
• As the Fig. 1.49 shows the duty of the application, transport, and network layers is end-
to-end.
• However, the duty of the data-link and physical layers is hop-to-hop, in which a hop is
a host or router. In other words, the domain of duty of the top three layers is the
internet, and the domain of duty of the two lower layers is the link.
Fig. 1.50: Identical Objects in the TCP/IP Protocol Suite
1.66
• Another way of thinking of the logical connections in TCP/IP network-model is to think

about the data unit created from each layer.
• In the top three layers, the data unit (packets) should not be changed by any router or
link-layer switch.
• In the bottom two layers of TCP/IP protocol suite, the packet created by the host is
changed only by the routers, not by the link-layer switches.
• Fig. 1.50 shows the identical objects below each layer related to each device.
1.4.2.2 Functions of Each Layer of TCP/IP Reference Model

• After understanding the concept of logical communication, we are ready to briefly dis-
cuss the duty/function of each layer in TCP/IP network model.
1. Physical Layer:
• The physical layer (lowest level) in TCP/IP network model is responsible for carrying
individual bits in a frame across the link.
• The communication between two devices at the physical layer is still a logical
communication because there is another, hidden layer, the transmission media, under
the physical layer.
• Two devices are connected by a transmission medium (cable or air). We need to know
that the transmission medium does not carry bits; it carries electrical or optical
signals.
• So the bits received in a frame from the data-link layer are transformed and sent
through the transmission media, but we can think that the logical unit between two
physical layers in two devices is a bit.
• There are several protocols that transform a bit to a signal, including Ethernet, Token
ring (820.3), FDDI, X.25, Frame relay.
2. Data Link Layer:
• An internet is made up of several links (LANs and WANs) connected by routers. There
may be several overlapping sets of links that a datagram can travel from the host to
the destination. The routers are responsible for choosing the best links.
• However, when the next link to travel is determined by the router, the data-link layer
in TCP/IP protocol suite is responsible for taking the datagram and moving it across
the link.
• The link can be a wired LAN with a link-layer switch, a wireless LAN, a wired WAN, or
a wireless WAN.
• We can also have different protocols used with any link type. In each case, the data-
link layer is responsible for moving the packet through the link.
• TCP/IP does not define any specific protocol for the data-link layer. It supports all the
standard and proprietary protocols.
1.67
• Any protocol that can take the datagram and carry it through the link suffices for the
network layer. The data-link layer takes a data-gram and encapsulates it in a packet
called a frame.
• Each link-layer protocol may provide a different service. Some link-layer protocols
provide complete error detection and correction, some provide only error correction.
• Examples of data-link layer protocols are Ethernet, IEEE 802.2 framing and Point-to-
Point Protocol (PPP) framing.
3. Network Layer:
• The network layer in TCP/IP protocol suite is responsible for creating a connection
between the source computer and the destination computer. The communication at
the network layer is host-to-host.
• However, since there can be several routers from the source to the destination, the
routers in the path are responsible for choosing the best route for each packet.
• The network layer is responsible for host-to-host communication and routing the
packet through possible routes.
• The network layer in the Internet includes the main protocol Internet Protocol (IP),
that defines the format of the packet, called a datagram at the network layer. IP also
defines the format and the structure of addresses used in network layers.
• IP is also responsible for routing a packet from its source to its destination, which is
achieved by each router forwarding the datagram to the next router in its path.
• IP is a connectionless protocol that provides no flow control, no error control, and no
congestion control services.
• The network layer in TCP/IP protocol suite also has some auxiliary protocols that help
IP in its delivery and routing tasks. Some of them are:
(i) The Internet Control Message Protocol (ICMP) helps IP to report some
problems when routing a packet.
(ii) The Internet Group Management Protocol (IGMP) helps IP in multitasking.
(iii) The Dynamic Host Configuration Protocol (DHCP) helps IP to get the network-
layer address for a host.
(iv) The Address Resolution Protocol (ARP) helps IP to find the link-layer address of
a host or a router when its network-layer address is given.
4. Transport Layer:
• The logical connection at the transport layer in TCP/IP protocol suite is also end-to-end.
The transport layer at the source host gets the message from the application layer,
encapsulates it in a transport-layer packet (called a segment or a user datagram in
different protocols) and sends it, through the logical (imaginary) connection, to the
transport layer at the destination host.
1.68
• In other words, the transport layer in TCP/IP protocol suite is responsible for giving
services to the application layer, to get a message from an application program
running on the source host and deliver it to the corresponding application program on
the destination host.
• There are a few transport layer protocols in the Internet, each designed for some
specific task. The main protocol of transport layer is Transmission Control Protocol
(TCP).
• TCP is a connection-oriented protocol that first establishes a logical connection
between transport layers at two hosts before transferring data. TCP creates a logical
pipe between two TCPs for transferring a stream of bytes.
• TCP provides flow control (matching the sending data rate of the source host with the
receiving data rate of the destination host to prevent overwhelming the destination),
error control (to guarantee that the segments arrive at the destination without error
and resending the corrupted ones), and congestion control to reduce the loss of
segments due to congestion in the network.
• The other common transport layer protocol is User Datagram Protocol (UDP), UDP is a
connectionless protocol that transmits user datagrams without first creating a logical
connection.
• In UDP, each user datagram is an independent entity without being related to the
previous or the next one (the meaning of the term connectionless).
• UDP is a simple protocol that does not provide flow, error, or congestion control. UDP’s
simplicity, which means small overhead, is attractive to an application program that
needs to send short messages and cannot afford the retransmission of the packets
involved in TCP, when a packet is corrupted or lost.
• A new protocol, Stream Control Transmission Protocol (SCTP) is designed to respond to
new applications that are emerging in the multimedia.
5. Application Layer:
• Fig. 1.49 shows the logical connection between the two application layers is end-to-
end. The two application layers’ exchange messages between each other as though
there were a bridge between the two layers. However, we should know that the
communication is done through all the layers.
• Communication at the application layer is between two processes (two programs
running at this layer). To communicate, a process sends a request to the other process
and receives a response.
• Process-to-process communication is the duty of the application layer. The application
layer in the Internet includes many predefined protocols, but a user can also create a
pair of processes to be run at the two hosts.
1.69
• The HyperText Transfer Protocol (HTTP) is a vehicle for accessing the World Wide
Web (WWW or Web). The Simple Mail Transfer Protocol (SMTP) is the main protocol
used in electronic mail (e-mail) service.
• The File Transfer Protocol (FTP) is used for transferring files from one host to another.
The Terminal Network (TELNET) and Secure Shell (SSH) are used for accessing a site
remotely.
• The Simple Network Management Protocol (SNMP) is used by an administrator to
manage the Internet at global and local levels.
• The Domain Name System (DNS) is used by other protocols to find the network-layer
address of a computer. The Internet Group Management Protocol (IGMP) is used to
collect membership in a group.
1.4.3 Comparison of OSI and TCP/IP Models (Oct. 17, 18; April 18, 19)
• Following table compare OSI reference model and TCP/IP model:
Sr.
OSI Reference Model TCP/IP Model
No.
1. OSI refers to Open Systems TCP refers to Transmission Control
Interconnection. Protocol/ Internet Protocol.
2. OSI model had Seven (7) layers. TCP/IP model has five (5) layers.
3. OSI is less reliable. TCP/IP is more reliable.
4. Developed by ISO (International Developed by the Department of Defense
Standard Organization). (DoD).
5. OSI is a conceptual model. TCP/IP is a client-server model, i.e. when
the client requests for service it is provided
by the server.
6. Protocol independent standard. Protocol dependent standard.
7. OSI follows a horizontal approach. TCP/IP follows a vertical approach.
8. OSI model follows a bottom-up TCP/IP follows a top-bottom approach.
approach.
9. In the OSI model, model was In the TCP/IP model, protocol were
developed before the development developed first and then the model was
of protocols. developed.
10. Model describes any type of Model only describes TCP/IP which is not
network. useful for describing any other networks.
1.70
Application
Presentation Application Several application

protocols
Session
Transport Transport Several transport

protocols
Network Network Internet protocol and

same helping protocols
Data link Data link

Underlying LAN and WAN
technology
Physical Physical
Fig. 1.51: Comparison Diagram for ISO-OSI and TCP/IP Network Reference/Models
1.4.4 Connection Devices in different Layers (April 17, Oct. 18)
• Computer network connecting devices are physical devices which are required for
communication and interaction between devices on a computer network.
• The network devices are playing an important role in network communication, each
device has a different role.
Network Connecting Devices for OSI Model:
• Fig. 1.52 shows positions of network devices in OSI model.
Fig. 1.52: OSI Model and Position of Networking Devices

• Hardware devices that are used to connect computers, printers, fax machines and
other electronic devices to a network are called network devices.
• Without network devices a computer network cannot be made and work. Following
are the list of network devices shows their need and importance in networking:
1. Hub: Hub is important because it broadcasts data from one port to all other ports
in the network. A hub works on the physical layer (Layer 1) of the OSI model. Hubs
are available with 4, 8, 12, 24, 48 ports. A hub is a networking device which
1.71
receives signal from the source, amplifies it and sends it to multiple destinations or
computers.
2. Switch: It is used in to send data point to point, meaning directly from one
computer to another through a switch. Network switch is a small hardware device
that joins multiple computers together within one Local Area Network
(LAN). Network switches operate at layer two (Data Link Layer) of the OSI
model. A network switch can be defined as the device that connects the network
devices or network segments. Switches available with 4, 8, 12, 24, 48, 64 ports.
3. Repeater: The repeaters are used in places where amplification of input signal is
necessary. A repeater (or regenerator) is an electronic device that operates on a
physical layer of the OSI model. A repeater is an electronic device that receives
a signal and retransmits it. Repeaters are used to extend transmissions so that the
signal can cover longer distances.
4. Router: A router is a networking device that forwards data packets between
computer networks. Routers perform the traffic directing functions on the
Internet. A router operates in the Network, Data link, and Physical layers of the
OSI model.
5. Gateway: Needed when two different network technologies are being used.
Gateway acts as translator switch-intelligent device which sends data to a
particular port. Gateway works on all seven (7) layers of the OSI model. Gateway is
a network device used to connect two or more dissimilar networks. In networking
parlance, networks that use different protocols are dissimilar networks.
6. Bridge: A network bridge is a computer networking device that creates a single
aggregate network from multiple communication networks or network segments.
This function is called network bridging. Bridging is distinct from routing. Routing
allows multiple networks to communicate independently and yet remain separate,
whereas bridging connects two separate networks as if they were a single
network. It works at the Data Link layer of the OSI Model and connects the
different networks together and develops communication between them. If one or
more segments of the bridged network are wireless, the device is known as
a wireless bridge.
7. Modem: Modem is a device that enables a computer to send or receive data over
telephone or cable lines. The modem works at the Physical Layer of the OSI model.
The main function of the modem is to convert digital signals into analog and vice
versa. Modem is a combination of two devices − modulator and demodulator.
The modulator converts digital data into analog data when the data is being sent
by the computer. The demodulator converts analog data signals into digital data
when it is being received by the computer.
1.72
Network Connecting Devices for TCP/IP Model:

• The TCP/IP model is based on a five-layer model for networking.
• Physical layer of TCP/IP model is responsible for physical connectivity of two devices.
Some of the devices used in Physical layer are, Ethernet, Hub, Repeaters etc.
• The devices used in the Data link layer of TCP/IP model are Bridges, Modem, Network
Interface Cards (NICs) etc.
• The devices used in the Network layer of TCP/IP model are Router, Brouter etc. The
devices used in transport layer of TCP/IP model are Gateway, Firewall etc.
• The devices used in the Application layer of TCP/IP model are Phones, Servers,
Gateways etc.
PRACTICE QUESTIONS
Q.I Multiple Choice Questions:
1. Which is refers to the exchange of data between two devices through some form of
wired or wireless transmission media?
(a) Data communication (b) Networking
(c) Data transfer (d) None of these
2. Which is an interconnection of computers and computing equipment using either
wires or ratio waves and can share data and computing resources?
(a) Data Communication (b) Computer Network
(c) Data Transmission (d) None of these
3. Which means the transfer of information between humans, computers or
machines in a meaningful way?
(a) Communication (b) Networking
(c) Data exchange (d) None of these
4. A network of networks is called an ______.
(a) Internetwork (b) Internet
(c) Both (a) and (b) (d) None of these
5. Components of data communication includes ______.
(a) Sender and Message (b) Protocol and Receiver
(c) Transmission Medium (d) All of these
6. Which plays an important role in networking?
(a) Data (b) Protocol
(c) Message (d) All of these
7. Data can be represented as ______.
(a) Text and Numbers (b) Audio and Video
(c) Images/Graphics (d) All of these
1.73
8. In which mode, the communication can take place in both directions, but only in
one direction at a time.
(a) Simplex (b) Full-duplex
(c) Half-duplex (d) All of these
9. Which is an interconnected collection of autonomous computers?
(a) Protocol (b) Network
(c) Communication (d) All of these
10. Network criteria includes ______.
(a) Performance (b) Reliability
(c) Type of communication media (d) All of these
11. Connection between two directly interconnected devices/nodes is referred to as
______ connection.
(a) Multipoint (b) Point-to-point
(c) Broadcast (d) All of these
12. Which refers to the way in which a network is laid out physically?
(a) Protocol (b) Message
(c) Topology (d) None of these
13. In which topology all nodes are connected to a central cable?
(a) Star (b) Bus
(c) Topology (d) None of these
14. Which is a privately-owned network covering a small geographical area, (less than
1 km) like a home, office etc.
(a) WAN (b) MAN
(c) LAN (d) All of these
15. Which is a combination of several point-to-point WANs that are connected by
switches.
(a) Point-to-point WAN (b) Switched WAN
(c) Multipoint MAN (d) All of these
16. A ______ needs to forward data from a network to another network when required.
(a) Switch (b) Hub
(c) Gateway (d) Router
17. In which network, a dedicated connection, (called a circuit) is always available
between the two end systems.
(a) Packet switched (b) Message switched
(c) Circuit switched (d) None of these
1.74
18. Which is the global system of interconnected computer networks that uses
the Internet protocol suite (TCP/IP) to communicate between networks and
devices.
(a) Internet (b) LAN
(c) WAN (d) MAN
19. Internet access is done by ______.
(a) Dial-up service (b) DSL Service
(c) Cable modem (d) All of these
20. Which is a conceptual layout in networking that describes how communication
between devices should occur.
(a) Reference model (b) Network model
(c) Both (a) and (b) (d) All of these
21. How many layers contains OSI model?
(a) 4 (b) 7
(c) 5 (d) 6
22. Design issues of the layer includes ______.
(a) Addressing and Routing (b) Error and Flow control
(c) Security and Scalability (d) All of these
23. How many layers contains TCP/IP model?
(a) 4 (b) 7
(c) 5 (d) 6
24. A ______ operates in the Network, Data link, and Physical layers of the OSI model.
(a) Hub (b) Router
(c) Switch (d) Repeater
ANSWERS
1. (a) 2. (b) 3. (a) 4. (c) 5. (d) 6. (a) 7. (d)
8. (c) 9. (b) 10.(d) 11. (b) 12. (c) 13. (b) 14. (c)
15. (b) 16. (a) 17. (c) 18. (a) 19. (d) 20. (c) 21. (b)
22. (d) 23. (c) 24. (b)
Q. II Fill in the Blanks:
1. A system of interconnected computers and computerized peripherals such as
printers is called computer ______.
2. ______ is the process of establishing connection or link between two entities for the
transfer/exchange of information.
1.75
3. The interconnected computers can share resources, which called ______.

4. The network ______ consists of instruction sets, that make possible the services that
we expect from a network.
5. ______ communication is the exchange of information between two computers
capable of generating processing and interpreting data.
6. The meaningful, logical and processed data is called as ______.
7. The ______ is the information (data) to be communicated.
8. A ______ is a set of rules that governs data communications.
9. In ______ mode, the communication can take place in both directions
simultaneously.
10. ______ enables its users to share and access enormous amount of information
worldwide.
11. A set of layers and protocols is called a network ______.
12. A ______ network is a set of devices connected by channels on links and provides a
service between users located at various geographical points.
13. Connection between two directly interconnected devices/nodes is referred to as
______ connection.
14. In ______ topology each device has a dedicated point-to-point link on it to a central
controller, usually called hub or switch.
15. In ring topology, the computers in the network are connected in a circular fashion
which form of a ______.
16. ______ are widely used to connect personal computers and workstations to share
resources (printers, scanners) and exchange information.
17. A ______ is a computer network that interconnects users with computer resources
in a geographic region of the size of a metropolitan area like entire city.
18. ______ oriented services are similar to telephone systems which are highly reliable.
19. The connection of two or more networks is called ______ or Internet.
20. ______ is an organization offering access to internet with monthly/yearly fee.
21. Digital Subscriber Line (DSL) provides Internet access by transmitting digital data
over the wires of a local ______ network.
22. In ______ connection, computer uses its modem to dial a telephone number given to
the user by an Internet Service Provider.
23. A communication system designed in such a manner is referred to as ______
architecture.
24. The ______ model is a layered framework for the design of network systems that
allows communication between all types of computer system.
1.76
25. The ______ is the conceptual model and set of communications protocols used on
the Internet and similar computer networks.
ANSWERS
1. Network 2. Communication 3. Networking 4. Software
5. Data 6. Information 7. Message 8. Protocol
9. Full-duplex 10. Internet 11. Architecture 12. Communication
13. Point-to-point 14. Star 15. Ring 16. LANs
17. MAN 18. Connection 19. Internetworks 20. Internet Service
Provider (ISP)
21. Telephone 22. Dial-up 23. Layered 24. OSI
25. TCP/IP
Q. III State True or False:
1. The process of sending or receiving data between two points/entities of a computer
networks is known as data communication.
2. A network is a set of devices (sometimes called nodes or stations) interconnected
by media links.
3. A data communication network is a collection of two or more computing devices
that are interconnected to enable them to share data.
4. A computer network is a set of electronically connected computers which can
share information and resources among themselves.
5. The hardware consists of the physical equipment that carries signals from one
point to the network to another.
6. A computer network is an interconnection of computers and computing
equipment using either wires or radio waves over small or large geographic area.
7. The term topology refers to the way a network is laid out, either physically or
logically.
8. Internet is computer based global information system which composed with
number of interconnected computer networks.
9. A data communication system has five components namely sender, message,
receiver, transmission medium, and protocol.
10. In reference model a layer is a grouping of related tasks involving the transfer of
information.
11. In full-duplex mode, the communication can take place in only one direction.
12. Computer network is divided in to wired and wireless network.
13. In point-to-point connection, a single link is shared by multiple devices.
1.77
14. In a mesh network topology, each of the network node, computer and other
devices, are interconnected with one another with dedicated point to point link.
15. A WAN is a geographically-dispersed collection of LANs.
16. The switched WAN connects the end systems.
17. An internetwork is a collection of individual networks, connected by intermediate
networking devices, that functions as a single large network.
18. In a computer network, the communication between the two ends is done in
blocks of data known as packets.
19. The Internet is the global system of interconnected computer networks that uses
the Internet protocol suite (TCP/IP) to communicate between networks and
devices.
20. A cable modem can be added to or integrated with a set-top box that provides the
TV set for Internet access.
21. The entities comprising the corresponding layers on different machines are called
peers.
22. A network consists of a series of levels called layers.
23. TCP/IP stands for Transmission Control Protocol/Internet Protocol.
24. Connection-oriented service is modeled after the postal system. In this type of
services, no connection is established between source and destination.
25. The seven layers of OSI model are physical layer, data link layer, network layer,
transport layer, session layer, presentation layer and application layer.
26. TCP/IP protocol suite also called as Internet protocol suite.
27. The original TCP/IP protocol suite was defined as four software layers built upon
the hardware. Today, however, TCP/IP is thought of as a five-layer model.
28. Data in networking can be available in the form of text, number, images, audio
and video.
29. Internet is a world-wide global system of interconnected computer networks.
30. Networking hardware is designed to help set up, manage, and/or monitor
computer networks.
ANSWERS
1. (T) 2. (T) 3. (T) 4. (T) 5. (T) 6. (T)
7. (T) 8. (T) 9. (T) 10. (T) 11. (F) 12. (T)
13. (F) 14. (T) 15. (T) 16. (T) 17. (T) 18. (T)
19. (T) 20. (T) 21. (T) 22. (T) 23. (T) 24. (F)
25. (T) 26. (T) 27. (T) 28. (T) 29. (T) 30. (F)
1.78
Q. IV Answer the following Questions:

(A) Short Answer Questions:
1. What is data communication?
2. List components of data communication.
3. What is network?
4. Define the term computer network.
5. Define LAN.
6. What is WAN?
7. List design issues of the layer.
8. Give function of network software.
9. Define switching.
10. What is protocol?
11. Explain purpose of reference model.
12. Define the term internet.
13. List layers of OSI model.
14. Define MAN.
15. What is topology?
16. What is meant by data representation?
17. List layers of TCP/IP model.
18. Which methods are used for accessing Internet.
19. State any two characteristics of data communication.
20. What network hardware?
21. List network devices for TCP/IP model.
22. What are the types of computer network?
(B) Long Answer Questions:
1. With the help of diagram describe components of data communication.
2. Explain computer network diagrammatically.
3. What is data and information? Define and compare them.
4. Describe modes in data communication with advantages and disadvantages.
5. Explain network criteria in detail.
6. Describe point-to-point and multipoint connections in detail.
7. Define topology. List types of topologies.
8. Describe the following topologies with their advantages and disadvantages:
(i) Bus
(ii) Tree
1.79
(iii) Mesh
(iv) Star
(v) Ring.
9. Compare LAN, MAN and WAN.
10. Explain point-to-point WAN and switched WAN with diagram.
11. What is circuit switched network? Explain in detail.
12. Define internet? Enlist ways for accessing internet.
13. Explain protocol hierarchies in detail.
14. Describe design issues of layers in detail.
15. With the help of diagram describe connection-oriented and connection-less
services.
16. Describe OSI model with its layers. Also explain functions of each layer.
17. Explain TCP/IP OSI model with its layers. Also explain functions of each layer.
18. Differentiate between OSI and TCP/IP models.
UNIVERSITY QUESTIONS AND ANSWERS

April 2016
1. Give diagrammatic representation of bus and mesh topology. [1 M]
Ans. Refer to Sections 1.2.4.2.2 and 1.2.4.2.4.
2. What are the advantages of point-to-point network? [2 M]
Ans. Refer to Section 1.2.4.1, Point (1).
3. What is internetworking? [1 M]
Ans. Refer to Section 1.2.5.4.
April 2017
1. Define protocol with its key elements. [1 M]
Ans. Refer to Section 1.1.2, Point (4).
2. Define mesh topology. [1 M]
Ans. Refer to Section 1.2.4.2.4.
3. Which devices operate at physical layer? [1 M]
Ans. Refer to Section 1.4.4.
4. State the difference between LAN and WAN. [5 M]
Ans. Refer to Page No. 1.29 and 1.30.
5. What are the responsibilities of session and presentation layer? [5 M]
Ans. Refer to Section 1.4.1.2, Points (5) and (6).
1.80
6. Explain star topology with their advantages. [2 M]

October 2017
1. List two similarities between TCP/IP and OSI model. [1 M]
2. For n devices in a network, what is the number of cables required for ring
topology? [1 M]
3. Explain the OSI reference model in detail. [5 M]
4. Write in detail about simplex, half duplex and full duplex data communication. [5 M]
April 2018
1. Write any two advantages of star topology. [1 M]
2. What is the responsibility of physical layer. [1 M]
3. Give the diagrammatic representation of mesh topology. [1 M]
Ans. Refer to Sections 1.2.4.2.4.
4. What is topology? Explain the ring topology with advantages and
disadvantages. [5 M]
Ans. Refer to Sections 1.2.4.2 and 1.2.4.2.3.
5. Compare and contrast OSI and TCP/IP model. [5 M]
October 2018
1. Define protocols. What are its key elements? [1 M]
2. Which topology requires a multipoint connection? [1 M]
3. Which device operates in physical layer? [1 M]
4. Explain similarities and differences between OSI and TCP/IP reference
models. [5 M]
1.81
5. State advantages and disadvantages of star topology. [2 M]

April 2019
1. Write disadvantages of star topology. [1 M]
2. List any four application layer protocols. [1 M]
3. What is computer network? [1 M]
4. Compare and contrast OSI and TCP/IP model. [5 M]
5. State advantages and disadvantages of mesh topology. [2 M]
❖❖❖
1.82
CHAPTER
2
Lower Layers
Objectives…
To understand Lower Layers (Physical and Data Link)
To learn Design Issues and Services of Data Link Layer
To study Framing Methods Channel Allocation Problem with its Methods
To understand Switching and its various Techniques
To study Wired LAN and Wireless LAN
2.0 INTRODUCTION (April 18)
• The physical layer (layer 1) and data link layer (layer 2) are the lower layers of the OSI
model. The physical layer defines the physical characteristics and functions of the
physical devices and interfaces so that transmission can occur.
• The data link layer is responsible for the reliable transfer of data frames from one
node to another connected by the physical layer.
• The physical layer is concerned with transmission of raw bits over a communication
channel/medium.
• The physical layer specifies the mechanical, electrical and procedural network
interface specifications and the physical transmission of bit streams over a
transmission medium connecting two pieces of communication equipment.
• To be transmitted, data must be transformed to electromagnetic signals. It is the layer
which actually interacts with transmission media, which connects network
components together.
• The physical layer provides services to the data link layer. The data in the data link
layer is organized in 0’s and 1’s in smaller frames. These streams of 0’s and 1’s must be
converted into signals. Physical layer does that.
• The data link layer is the second lower layer in the OSI model, above the physical
layer, which ensures that the error free data is transferred between the adjacent
nodes in the network.
2.1
Computer Networks – I (S.Y.B.Sc. Comp. Science) Lower Layers
2.1 COMMUNICATION AT THE PHYSICAL LAYER

• The physical layer (layer 1) of OSI and TCP/IP models deals with transmission of
individual bits from one node to another over a physical medium.
• Communication at the physical layer means exchanging signals. Signal is an electrical,
electronic or optical representation of data, which can be sent over a communication
medium.
• Data is transmitted from one point or entity to another point or entity by means of
electrical signals that may be in digital and analog form.
• Data refers to information that conveys some meaning based on some mutually agreed
up rules or conventions between a sender and a receiver and today data comes in a
variety of forms such as text, graphics, audio, video and animation.
• Signals can be either analog or digital. Analog signals have infinite values in a range
and Digital signals have a limited number of defined values.
• Analog signals are used to represent analog data, and digital signals are used to
represent digital data.
• Data can be analog or digital. Analog data refers to information that is continuous.
For example, sounds made by a human voice generate continuous waves in the air.
• Digital data refers to information that has discrete states. Digital data takes on discrete
values. For example, data is stored in computer memory in the form of 0s and 1s.
• The purpose of the physical layer is to create the electrical, optical, or microwave
signal that represents the bits in each frame. These signals are then sent on the media
one at a time.
Signals on the physical media

Fig. 2.1: Communication at the Physical Layer of OSI Model
2.2
• It is also the job of the physical layer to retrieve these individual signals from the
media, restore them to their bit representations, and pass the bits up to the data link
layer as a complete frame.
• The OSI physical layer provides the means to transport across the network media the
bits that make up a data link layer frame.
• The physical layer accepts a complete frame from the data link layer and encodes it as
a series of signals that are transmitted onto the local media.
Fig. 2.2: Communication at the Physical Layer of TCP/IP Model

2.3
2.1.1 Data Rate Limits (April 16, 19, Oct. 18)
• In data communication, a very important consideration is how fast we can send data,
in bits per second, over a channel. Data rate depends upon:
1. The bandwidth available,
2. The level of the signals we use, and
3. The quality of the channel (the level of noise).
• The maximum rate at which data can be correctly communicated over a channel in
presence of noise and distortion is known as its channel capacity.
• Two formulas were developed to calculate the data rate: one by Nyquist for a noiseless
channel, another by Shannon for a noisy channel.
Nyquist Bit Rate Formula for Noiseless Channel:
• For a noiseless channel, the Nyquist bit rate formula defines the theoretical maximum
bit rate.
Bit rate = 2 × bandwidth × log2L
• Bandwidth is the bandwidth of the channel. L is the number of signal levels used to
represent data. Bit rate is the bit rate in bits per second.
• Practically, there is a limit on bit rate. Increasing the levels of a signal may reduce the
reliability of the system.
Example 1: Consider a noiseless channel with a band width of 4000 Hz transmitting a
signal with two signal levels. What will be the maximum bit rate?
Solution:
Bit Rate = 2 × bandwidth × log2 L
= 2 × 4000 × log22
= 8000 bps.
Example 2: We need to send 265 kbps over a noiseless channel with a bandwidth of 20
kHz. How many signal levels do we need?
Solution:
Bit Rate = 2 × bandwidth × log2L
265,000 = 2 × 20,000 × log2L
Log2L = 6.625
6.625
L =2
L = 98.7 levels
Example 3: Let us consider the telephone channel having bandwidth B = 4 kHz.
Assuming there is no noise, determine channel capacity for the encoding levels-128.
2.4
Solution:
Bit Rate = 2 × bandwidth × log2L
= 2 × 4000 × log2128
= 8000 × 7 = 56 Kbits/s
Shannon’s Law for Noisy Channel: (Oct. 17)
• In reality, we cannot have a noiseless channel, the channel is always noisy. When
there is noise present in the medium, the limitations of both bandwidth and noise
must be considered.
• A noise spike may cause a given level to be interpreted as a signal of greater level, if it
is in positive phase or a smaller level, if it is negative phase. Noise becomes more
problematic as the number of levels increases.
• In 1994, Shannon introduced a formula, called Shannon capacity, to determine the
theoretical highest data rate for noisy channels.
Capacity = Bandwidth × Log2 (1 + SNR)
• Bandwidth is bandwidth of the channel. SNR is signal to noise ratio; capacity is the
capacity of the channel in bits per second.
• Signal to noise ratio (SNR) is calculated as,
Average signal power
SNR = Average noise power
Example 4: Calculate the theoretical highest bit rate of a regular telephone line. A
telephone line normally has a bandwidth of 3000 Hz assigned for data communication.
The signal to noise ratio is usually 3162. Calculate the capacity.
Solution: C = B log2 (1 + SNR)
= 3000 log2 (1 + 3162)
= 3000 log2 3163
= 3000 × 11.62
Capacity = 341860 bps
Example 5: The digital signal is to be designed to permit 160 kbps for a bandwidth of
20 KHz. Determine (a) number of levels and (b) S/N ratio.
Solution:
(a) Apply Nyquist Bit Rate to determine number of levels.
C = 2B log2(L),
3 3
160 × 10 = 2 × 20 × 10 log2(L)
log2(L) = 4
4
L = 2 , which means 4bits/baud.
2.5
(b) Apply Shannon capacity to determine the S/N ratio:

C = B log2(1 + S/N),
3 3
160 × 10 = 20 × 10 log2(1 + S/N)
log2(1 + S/N) = 8
8
S/N = 2 - 1
S/N = 255
S/N = 24.07 dB.
Example 6: Given a channel with an intended capacity of 20 Mbps. The bandwidth of
the channel is 3MHz. What signal-to-noise ratio is required in order to achieve this
capacity?
Solution:
According to Shannon’s Capacity formula, the maximum channel capacity (in bps) is
given by the equation:
C = B log2(1 + SNR)
Where B is the bandwidth and SNR is the signal-to-noise ratio.
6 6
Given B = 3 MHz = 3 × 10 Hz, and C = 20 Mbps = 20 × 10 bps,
So,
6 6
20 × 10 = 3 × 10 log2 (1 + SNR)
log2 (1 + SNR) = 20 / 3 = 6.667
1 + SNR = 102
Hence, SNR = 101
Example 7: What is the channel capacity for a tele-printer channel with a 300 Hz
bandwidth and a signal-to-noise ratio of 3 DB?
Solution:
Using Shannon's equation: C = B log2(1 + SNR)
We have B = 300 Hz and SNR (in dB) = 3,
0.3
Therefore, SNR = 10
C = 300 log2(1 +100.3)
C = 300 log2(2.995)
C = 474 bps
2.1.2 Performance of Network

• One important issue in networking is the performance of the network. Performance of
the network depends upon several factors like Bandwidth, Throughput, Latency
(delay), Bandwidth-delay product and Jitter.
2.6
2.1.2.1 Bandwidth
• In computer networks, the term bandwidth refers to the speed of data transmissions.
It is a measure of the data that can be transmitted from one point to another in a given
amount of time.
• To measure network performance, the one characteristic is bandwidth. In networking,
we use the term bandwidth in two contexts with following two different measuring
values:
1. Bandwidth in Hertz (Hz): Bandwidth in hertz refers to the range of frequencies in
a composite signal or the range of frequencies that the channel can pass. For
example, the bandwidth of a subscriber telephone line is 4kHz.
2. Bandwidth in bits per second (bps): Bandwidth in bits per second refers to the
speed of bit transmission in a channel or link. For example, the bandwidth of a
Fast Ethernet network or the links is a maximum of 100 Mbps means this network
can send 100 Mbps.
• There is an explicit relationship between in hertz and bandwidth in bits per seconds.
An increase in bandwidth in hertz means an increase in bandwidth in bits per second.
2.1.2.2 Throughput
• The throughput is a measure of how fast we can actually send data through a network.
Throughput is the rate of successful message delivery over a communication
channel/medium.
• Bandwidth in bits per second and throughput seems to be the same, but they are
different. Throughput is controlled by available bandwidth.
• A link may have bandwidth of B bps, but we can only send T bps through this link with
T always less than B. Bandwidth is a potential measurement of a link and throughput
is an actual measurement of how fast we can send data.
• Consider a highway designed to transmit 1000 cars per minute from one point to
another. However, if there is congestion on the road, this can be reduced to 200 cars
per minute. So bandwidth is 1000 cars per minute and throughput is 200 cars per
minute.
Example 8: A network with bandwidth of 10 Mbps can pass only an average of 12,000
frames per minute with each frame carrying an average of 10,000 bits. What is the
throughput of this network?
Solution: Throughput is the amount of data moved successfully from one place to
another place. That is 12,000 frames per minute and each Frame is carrying 10,000 bits.
2.7
12,000 × 10,000
Throughput = 60
= 2 × 1000000 bits/seconds
= 2Mbps
2.1.2.3 Latency (April 17)
• The latency or delay defines how long it takes for an entire message to completely
arrive at the destination from the time the first bit is sent out from the source.
• Latency is the time it takes for a packet to get across the network, from source to
destination.
Latency = Propagation time + Transmission time + Queuing time + Processing delay
Propagation Time:
• Propagation time measures the time required for a bit to travel from the source to
destination.
Distance between Two Points
Propagation time = Propagation speed
Example 9: What is the propagation time if the distance between the two points is
8
12,000 km? Assume the propagation speed to be 2.4 × 10 m/s in cable.
Solution: We calculate the propagation time as,
12,000 × 1000
Propagation time = 2.4 × 108
= 50 ms
Transmission Time:
• In data communication, a message containing bits is sent. The first bit may take a time
equal to the propagation time to reach its destination; the last bit also may take the
same amount of time.
• However, there is a time between the first bit leaving the sender and the last bit
arriving at the receiver. The first bit leaves earlier and arrives earlier, the last bit
leaves later and arrives later. The time required for transmission of a message
depends on the size of the message and the bandwidth of the channel.
Message size
Transmission time = Bandwidth
Example 10: What is the propagation time for a 2.5 Kbyte message if the bandwidth of
the network is 1Gbps? Assume that the distance between the sender and the receiver is
8
12,000 km and that light travels at 2.4 × 10 m/s.
2.8
12000 × 1000
Solution: Propagation time = 2.4 × 10
8 = 50 ms
2500 × 8
Transmission Time = 10
9 = 0.020 ms
Queuing Time:
• The third component in latency is the queuing time, the time needed for each
intermediate or end device to hold the message before it can be processed.
• The queuing time is not fixed; it depends upon the load on the network. When there is
heavy traffic on the network, the queuing time increases.
2.1.2.4 Bandwidth-Delay Product
• By bandwidth and delay, we measure the performance of a network. In data
communication, the product of bandwidth and delay is very important.
• The bandwidth-delay product defines the number of bits that can fill the network link.
It gives the maximum amount of data that can be transmitted by the sender at a given
time before waiting for acknowledgment.
• Let us elaborate this issue, using two hypothetical cases as examples.
Case 1:
• Let us consider that we have a link with a bandwidth of 1bps and the delay of the link
is 5s. From the Fig. 2.3 we can say that the bandwidth delay product is 1 × 5, which is
the maximum number of bits that can fill the link. There can be no more than 5 bits at
any time on the link.
Fig. 2.3: Filling the Link with Bits for Case 1

Case 2:
• Now, assume we have a bandwidth of 4bps. Fig. 2.4 shows there can be
4 × 5 = 20 bits on the line. At each second, there are 4 bits on the line, the duration of
each bit is 0.25s.
2.9
Fig. 2.4: Filling the Link with Bits in Case 2
• The above two cases show that the product of bandwidth and delay is the number of
bits that can fill the link.
2.1.2.5 Jitter
• Another performance issue that is related to delay is jitter. Jitter is the variance in time
delay in milliseconds (ms) between data packets over a network. It is a disruption in
the normal sequence of sending data packets.
• In technical terms, jitter is a “packet delay variance”. The jitter is considered as a
problem when different packets of data face different delays in a network and the
data at the receiver application is time-sensitive, i.e. audio or video data.
• If real-time data (audio/video) on a packet switched network requires the preservation
of the time relationship between packets of a session.
• For example, consider a server sending 3 packets to a client. Every packet contains 10
sec video information.
• The first packet starts at 00.00.00, the second at 00.00.10 and the third at 00.00.20. Also
consider 1s is required for every packet to reach upto destination. The receiver can
play the first packet at 00:00:01, the second at 00:00:11 and the third at 00:00:21.
• Fig. 2.5 shows this idea.
2.10
Fig. 2.5: Noise
• But if the first packet arrives at 00:00:01 (1s delay), the second arrives at 00:00:15 (5s
delay), and the third arrives at 00:00:27 (7s delay) the receiver is not able to play
packets.
• After playing the first packet, he has to wait for the second. There is a gap between the
first and second packets and between second and third as the video is viewed at the
remote site.
• This concept is called jitter and shown in Fig. 2.6.
Fig. 2.6: Jitter
2.2 DATA LINK LAYER

• The data link layer is layer number 2 in the ISO-OSI and TCP/IP model. The data link
provides for the transfer of data frames between hosts connected to the physical link.
2.11
• Data link layer transforms the physical layer, a raw transmission facility to a reliable
link. It is responsible for moving frames from one hop (node) to the next i.e. hop-to-
hop delivery.
• The data link layer takes services from the Physical layer and gives services to the
network layer. Framing is a function of the data link layer. It provides a way for a
sender to transmit a set of bits that are meaningful to the receiver.
• Data link layer provides the mechanism of error control in which it detects and
retransmits damaged or lost frames.
• The data link layer converts the raw transmission facility provided by the physical
layer to a reliable and error-free link.
• The flow control function of data link layer coordinates that amount of data that can
be sent before receiving acknowledgement.
• The data link layer also supports access control function. When a single
communication channel is shared by multiple devices, the MAC sub-layer of the data
link layer helps to determine which device has control over the channel at a given
time.
• Data link layer has two sub-layers:
1. Logical Link Control (LLC): It deals with protocols, flow-control, and error
control
2. Media Access Control (MAC): It deals with actual control of media.
2.2.1 Design Issues

• The data-link layer is located between the physical and the network layers. The data
link layer provides services to the network layer; it receives services from the physical
layer.
• The design issues of the data link layer directly deal with its functionality and services.
The main functions of the data link layer are:
1. Controlling over the flow of data.
2. Providing the service interface to the network layer.
3. Dealing with the transmission errors.
• To do the above functions, the data link layer takes packets from the network layer
and converts them to frames for transmission.
• The data link layer divides the stream of bits received from the network layer into
manageable data units called as frames.
2.12
• The following are the some of the important design issues of the data link layer:
1. Services provided to the Network Layer: The data link layer acts as a service
interface to the network layer. How to provide a well-defined service interface in
the network layer on source machine to the network layer on destination machine.
2. Frame Synchronization/Framing: Frame synchronization is the major issue of
data link layer. The source machine sends data in blocks called frames to be the
destination machine. The starting and ending of each frame should be identified so
that the frame can be recognized by the destination machine. This design issue
determines how the bits of the physical layer are grouped into frames.
3. Flow Control: It deals with how to regulate the flow of frames so that slow
receivers are not swamped by fast senders.
4. Error Control: This design issue deals with transmission errors. It must provide
an error control mechanism to detect and retransmit damaged, duplicate, or lost
frames from source to destination.
2.2.2 Services
• Data link layer provides several services to the network layer. The one of the major
services provided is transferring the data from the network layer on the source
machine to the network layer on the destination machine.
• The source machine data link layer receives the data from the network layer and the
destination machine passes on this data to the network layer as shown in Fig. 2.7.
(a) Virtual Communication (b) Actual Communication
Fig. 2.7
2.13
• The path shown in Fig. 2.7 (a) is the virtual path. But the actual path is Network layer
-> Data link layer -> Physical layer on source machine, then to physical media and
thereafter physical layer -> Data link layer -> Network layer on destination machine.
• The data link layer services differ from system to system. The three main services are
Unacknowledged connectionless service, Acknowledged connectionless service and
Acknowledged connection oriented service.
1. Unacknowledged Connectionless Service:
• In this service, the sender machine sends the frames to the destination machine
without having the destination machine acknowledge them.
• No logical connection is established between sender and receiver beforehand. If the
frame is lost, no attempt is made to detect the loss or recover from it in the data link
layer.
• This is best suited when the error rate is very low, so the error recovery is left to
higher layers. It is best suited for real-time traffic, in which late data is worse than bad
data.
• Most LAN uses unacknowledged connectionless service.
2. Acknowledged Connectionless Service:
• In this service, no logical connection is used, but each frame sent is individually
acknowledged.
• Sender knows each frame has reached safety or not. In case of any error, it can be sent
again.
• This is used for unreliable channels, such as wireless systems.
3. Acknowledged Connection Oriented Service:
• Here, source and destination machines establish a connection before any data is
transferred. Each frame sent in a numbered and data link layer guarantees that each
frame sent is received.
• It also guarantees that each frame is received exactly once and that all frames are
received in the right order. Connection oriented service provides a reliable bit stream.
• Connection-oriented has three distinct phases:
(i) The connection is established when some initialization takes place.
(ii) The actual transmission of one and more frames.
(iii) The connection is released freeing the variables, buffers and other resources.
2.2.3 Framing (April 17, 18; Oct. 18)
• Framing is an important function of the data link layer. Group of physical layer bits
stream into units (messages) called frames. Breaking the bit stream into frames is
called framing.
2.14
• Data link layer is intermediate between the network layer and the physical layer. So
the data link layer must use the services provided by the physical layer.
• Physical layer accepts raw bit stream which is not guaranteed to be error free, and
attempts it to deliver it to destination.
• So it is the responsibility of the data link layer to detect errors such as the number of
bits received may be less or contain different values etc.
• The data link layer breaks the bit stream into the frames and computes checksum for
each frame. When frame comes at destination, the checksum is recomputed, if it is
different an error has occurred and it will be corrected.
• One method to achieve this framing is to insert time gaps between frames. But this
method is not useful because networks rarely male any guarantees about timings, so
these gaps might be squeezed or other gaps might be inserted during transmission. So
other methods have been devised.
• A frame is "the unit of transmission in a link layer protocol, and consists of a link layer
header followed by a packet.
• Framing in the data link layer separates a message from one source to a destination by
adding a sender address and a destination address.
• The destination address defines where the packet is to go; the sender address helps the
recipient acknowledge the receipt.
Sending machine Receiving machine
Packet Packet
Frame
Header Payload filed Trailer Header Payload filed Trailer
Fig. 2.8
• At the data link layer, it extracts messages from sender and provides it to receiver by
providing sender’s and receiver’s address.
• The advantage of using frames is that data is broken up into recoverable chunks that
can easily be checked for corruption.
Parts of a Frame:
• A frame has the following parts, (See Fig. 2.9):
1. Frame Header: It contains the source and the destination addresses of the frame.
2. Payload Field: It contains the message to be delivered.
2.15
3. Trailer: It contains the error detection and error correction bits.

4. Flag: It marks the beginning and end of the frame.
Flag Header Payload Field Trailer Flag
Fig. 2.9: Parts of Frame
2.2.4 Flow Control (Oct. 18)
• Consider a situation, the sender is running on a fast computer (or lightly loaded) and
the receiver is running on a slow (or heavily loaded) computer.
• The sender keeps on sending the frames at high speed, but the receiver is not able to
accept it.
• After some time, the receiver is totally swamped out. Even though the transmission
line is error free, at a certain point the receiver will be unable to handle frames and
start losing them.
• To prevent such types of situations flow control mechanisms are used by the data link
layers.
• Flow control is a technique for ensuring that a transmitting entity does not flood the
receiving entity with data.
• The data link layer regulates flow control so that a fast sender does not drown a slow
receiver.
• Flow control refers to a set of procedures used to restrict the amount of data the
sender can send before waiting for acknowledgment.
• The two flow control mechanisms are Feedback-based flow control, and Rate-based
flow control.
1. In feedback-based flow control, the receiver sends back information to the
sender giving it permission to send more data or at least telling the sender how the
receiver is doing.
2. In rate-based flow control, the protocol has a built-in mechanism that limits the
rate at which senders may transmit data, without using feedback from the
receiver.
2.2.5 Error Control

• Once the start and end of a frame is detected, the next problem is how to make sure
that all frames sent by the sender are delivered to the network layer of the destination
and in the proper order without error.
• Error control refers to a set of procedures that ensure that all the frames have
eventually been received at the destination.
2.16
• If any frames are lost or damaged during transmission, the error control mechanism
allows the receiver to inform the sender about it and coordinates the re-transmission
of those frames by the sender.
• The data link layer ensures error free link for data transmission. The issues it caters to
with respect to error control are:
1. Dealing with transmission errors.
2. Sending acknowledgement frames in reliable connections.
3. Retransmitting lost frames.
4. Identifying duplicate frames and deleting them.
5. Setting timer as soon as the frame is sent (if the acknowledgment does not receive
in the specified time, the same frame is retransmitted).
• If the acknowledgment does not reach before the timer runs out there will duplicate
copies of frames at the receiver end.
• To prevent this, generally a sequence number is assigned to outgoing frames. So the
receiver can distinguish the retransmission.
2.2.6 Congestion Control

• A congestion is a state occurring in a network layer when the message traffic is so
heavy that it slows down network response time.
• Although a link may be congested with frames, which may result in frame loss, most
data-link-layer protocols do not directly use a congestion control to alleviate
congestion, although some wide-area networks do.
• In general, congestion control is considered an issue in the network layer or the
transport layer because of its end-to-end nature.
2.2.7 Link Layer Addressing

• The other issue about the data link layer is the link-layer addresses. An IP addresses as
the identifiers at the network layer that define the exact points in the Internet where
the source and destination hosts are connected.
• However, in a connectionless internetwork such as the Internet we cannot make a
datagram reach its destination using only IP addresses.
• The reason is that each datagram in the Internet, from the same source host to the
same destination host, may take a different path.
• The source and destination IP addresses define the two ends but cannot define which
links the datagram should pass through.
• We need to remember that the IP addresses in a datagram should not be changed. If
the destination IP address in a datagram changes, the packet never reaches its
destination.
2.17
• If the source IP address in a datagram changes, the destination host or a router can
never communicate with the source if a response needs to be sent back or an error
needs to be reported back to the source.
• The above discussion shows that we need another addressing mechanism in a
connectionless internet the link layer addresses of the two nodes.
• A link layer address is sometimes called link address, sometimes a physical layer, and
sometimes a MAC address.
• Since, a link is controlled at the data link layer, the addresses need to belong to the
data link layer.
• When a datagram passes from the network layer to the data link layer, the datagram
will be encapsulated in a frame and two data link addresses are added to the frame
header.
• These two addresses are changed every time the frame moves from one link to
another link.
2.3 FRAMING METHODS (April 16, 18; Oct. 18)
• The framing methods are Character count, Flag bytes with byte stuffing, Starting and
ending flags, with bit stuffing and Physical layer coding violations.
2.3.1 Character Count

• Character count method uses a field in the header to specify the number of characters
in the frame.
• The data link layer at destination checks the character count, it knows how many
characters follow and where the end of frame is.
• Fig. 2.10 shows four frames of sizes 5, 5, 8 and 8 characters respectively.
(a) Without Error
(b) With One Error

Fig. 2.10: A Character Stream
2.18
• The problem with this method is the count can be garbled by a transmission error. In
Fig. 2.10 (b), the character 5 in frame 2 becomes 7, so destination frames get out of
synchronization and are unable to locate the start of the next frame. So in case of any
error the entire data has to be retransmitted.
2.3.2 Flag Bytes with Byte Stuffing
• The second framing method is flag byte. It solves the problem of resynchronization.
• Every frame starts and ends with a special bit pattern called a flag byte [011111110].
The flag byte is used as starting and ending delimiter as shown in Fig. 2.11.
(a) A Frame Delimited by Flag Bytes
(b) Four examples of Byte Sequences
Fig. 2.11
• If the receiver loses synchronization, it can just search the FLAG byte to find the end of
the current frame. Two consecutive flag bytes indicate the end of one frame and start
and next frame.
• A problem here comes with the binary data or float point numbers are being
transmitted. One way to solve this, a special ESC byte is inserted just before each
accidental flag byte in the data.
• The data link layer on the receiving end removes the escape byte before the data are
given to the network layer. This is known as byte stuffing or character stuffing.
• A major drawback is it uses 8-bit characters. Not all character codes use 8-bit
characters. Unicode uses 16-bit characters. So, it cannot be used for arbitrary sized
characters.
2.3.3 Starting and Ending Flags with Bit Stuffing (April 16)
• Here, it allows character codes with an arbitrary number of bits per character. Each
frame begins and ends with a special bit pattern, 01111110.
2.19
• Whenever, the sender's data link layer-encounters a five consecutive 1s in the data, it
automatically stuffs a 0-bit into the outgoing bit stream. So this is a bit stuffing.
• When the receiver sees 5 consecutive 1-bit followed by a 0-bit, it automatically de-
stuffs the 0-bit. If the user data contain the flag pattern, 01111110, this is transmitted
as 011111010 but stored in the receiver's memory as 01111110.
• The Fig. 2.12 shows the bit of stuffing.
Fig. 2.12: (a) Original Data (b) The Data appear on the Line
(c) The Data at Receivers Site
2.3.4 Physical Layer Coding Violations

• This is mainly used for the network in which the encoding contains some redundancy.
Some LANs encode 1-bit by using 2 physical bits.
• A 1-bit is a high-low pair and a 0-bit is a low-high pair. Every bit has a transition in the
middle. So it is easier for the receiver to locate the bit boundaries.
• Many Data link layer protocols use a combination of a character count with one of the
other methods for safety.
• When the frame arrives the count field shows end of the frame. If frame delimiter is
present at that position and checksum is correct the frame is accepted as valid,
otherwise the input stream is scanned for the next delimiter.
2.4 CHANNEL ALLOCATION METHODS

• Data link layer is divided into two sub layers i.e., Logical Link Control (LLC) and
Medium Access Control (MAC) Layer.
• The LLC provides addressing and control of the data link. MAC may refer to the
sublayer that determines who is allowed to access the media at any one time.
• A MAC layer provides addressing and channel access control mechanisms that make it
possible for several stations to communicate within a multiple access network that
incorporates a shared medium for example, Ethernet.
• Channel allocation is a process in which a single channel is divided and allotted to
multiple users in order to carry user specific tasks.
• The channel allocation problem shows how to allocate a single broadcast channel
among several competing users.
2.20
• The channel might be a portion of the wireless spectrum in a geographic region, or a

single wire or optical fiber to which multiple nodes are connected.
• Channel Allocation may be done using two ways Static Channel Allocation and
Dynamic Channel Allocation.
2.4.1 Static Channel Allocation

• In static channel allocation, a fixed portion of the frequency channel is allotted to each
competing user. This scheme is also referred to fixed channel allocation or fixed
channel assignment.
• The traditional way of allocation the single channel for multiple users is the Frequency
Division Multiplexing (FDM).
• If there are N users, the bandwidth is divided into N equal-sized portions, with each
user being assigned one portion. Since each user has a private frequency band, there
is no interference among users.
• When there is only a small and fixed number of users then FDM is a simple and
efficient allocation mechanism.
• However, when the number of senders is large and continuously varying then the
FDM is not suitable in those situations.
• If the total bandwidth is divided into N regions for N users and if fewer than N users
are communicating, then a large piece of valuable bandwidth is wasted.
• If more than N users want to communicate, some of them will not get the permission
for lack of bandwidth.
2.4.2 Dynamic Channel Allocation

• In dynamic channel allocation, frequency bands are not permanently assigned to the
users. Instead channels are allotted to users dynamically as needed, from a central
pool.
• Instead of reserving a special frequency band for every user and limiting the number
of users, the channel is allocated dynamically which can be used to transmit the data
for any number of users and at any time.
• For dynamic channel allocation, we will make some assumptions:
1. Station Modem: This modem consists of N independent stations, each with a user
that generates the frame of transmission. Once the frame has been generated, the
station is blocked and does nothing until the frame has successfully transmitted.
2. Single Channel Assumption: A single channel is available for all communication
i.e. All stations can transmit or receive from it.
2.21
3. Collision Assumption: If two frames are transmitted simultaneously, they may

overlap and result in the garbled signal. This event is called a collision. All
stations can detect collision and the collided frames are retransmitted.
4. Continuous Time: The frame transmission can begin at any instant and there is
no master clock dividing the time into discrete intervals.
5. Slotted Time: The time is divided into discrete intervals, slots. The frame
transmission always begins at the start of the time slot. One-time slot may have
contained transmission of many frames.
6. Carriers Sense Assumption: The station can tell if the channel is in use before
trying to use it. If the channel is sensed as busy, then no station will attempt to use
it until it becomes idle.
7. No Carriers Sense: Stations cannot sense the channel before trying to use it.
Instead it just transmits and after some time determines whether or not the
transmission was successful.
• The dynamic channel allocation is done considering a number of parameters so that
transmission interference is minimized.
2.4.3 Media Access Methods
• Media access methods are implemented at the data-link layer. When nodes are
connected and use a common link known as multipoint link or broadcast link, we
need a multiple access protocol to coordinate access to the link.
• A channel access method or multiple access method allows more than two terminals
connected to the same transmission medium to transmit over it and to share its
capacity.
• Number of protocols have been divided to handle access to a shared link and
categories into three groups as shown in Fig. 2.13.
Multiple - access
Protocols
Random Access Controlled Access Channelization

Protocols Protocols Protocols
ALOHA Reservation FDMA
Polling TDMA
CSMA/CD Token Passing CDMA
CSMA/CA
Fig. 2.13: Taxonomy of Multiple Access Protocols

2.22
• Protocols belonging to each group are shown in the Fig. 2.13.

1. Random Access Protocols: In these types of protocols any station can transmit at
any time. Use of the channel is not controlled by any station.
2. Controlled Access Protocols: In controlled access some sort of mechanism is used
to decide which station can transmit.
3. Channelization: In channelization, the available channel bandwidth is shared
either in frequency, time or code.
2.4.3.1 Random Access Protocols (April 17, 18, Oct. 17)
• In random access or contention methods, each station in the network has equal rights.
• In random access no station is superior to other stations and none station permits, or
does not permit another station to send.
• All can have direct access to the medium through which the information or data flows.
And that individual station is not controlled by any other station.
• At any instance of time, a station which wants to transmit data uses a procedure
defined by protocol, and decides whether to transmit or not.
• This decision depends upon whether the medium is free or busy. Random access
protocols have following two features:
1. There is no scheduled time for a station to transmit. Transmission is random
among the stations.
2. No rules specify which station should send next. Stations compete with one
another to access the medium. So these protocols are also called contention
methods.
• Even though, if more stations tried to send frames then there may be a conflict called
collision occurs or the frames may be destroyed or modified.
• To avoid this collision, modification etc., we need a procedure which handles the
situation properly. And it will help us to get answers of few questions like:
1. When can a station access the medium?
2. If the medium is busy, what station can do it?
3. If a collision occurs, what station can do it?
4. How does a station get information about the success and failure of a
transmission?
• To answer these types of questions, we have evolved of random access methods. These
are shown in Fig. 2.14.
• In this case, Multiple Access (MA) is a simple procedure called ALOHA. This method
was improved with the addition of procedure on the basis of sense of the medium
before transmitting. This is called CSMA i.e. carrier sense multiple access which has
two parallel methods i.e. CSMA/CD and CSMA/CA.
2.23
• CSMA/CA is for collision avoidance which defines a procedure to avoid the collision.
CSMA/CD is for collision detection which defines to be followed if a collision is
detected.
Fig. 2.14: Evolution of Random Access Methods

1. ALOHA: (April 16, Oct. 17)
• ALOHA is a system for coordinating and arbitrating access to a shared communication
Networks channel.
• ALOHA was developed in the 1970s by Norman Abramson and his colleagues at the
University of Hawaii to solve the problem of channel allocation.
• ALOHA was designed for a wireless LAN, but it can be used on any shared medium.
• The medium is shared between the stations. When a station sends data, another
station may attempt to send data at the same time. It is obvious that the data from the
two stations collide and are destroyed.
• There are two types of ALOHA protocols i.e. Pure ALOHA and Slotted ALOHA.
Pure ALOHA: (April 16, Oct. 17)
• The original ALOHA protocol is called pure ALOHA. Concept of ALOHA is very simple;
each station sends a frame whenever it has a frame to send.
• Since there is only one channel to share, there is possibility of collisions between
frames from different stations.
• Fig. 2.15 shows an example of frame collisions in pure ALOHA.
• In Fig. 2.15 as an example four stations are shown. Every station sends frames on a
shared medium.
• Some of these frames collide because multiple frames are in contention for the shared
channel. Only two frames, frame 1.1 and frame 2.2 survive. All other frames are
destroyed.
2.24
• In pure ALOHA, whenever any station transmits a frame, it expects the

acknowledgement from the receiver.
Collsion Collsion
Duration Duration
Frame 1.1
Frame 1.2 Frame 1.3
Time
Station 1
Frame 2.1 Frame 2.2

Time
Station 2
Frame 3.2
Frame 3.1
Time
Station 3 Frame 4.1
Time
Station 4
Fig. 2.15: Frames in Pure ALOHA
• If acknowledgement is not received within specified time, the station assumes that the
frame (or acknowledgement) has been destroyed.
• If the frame is destroyed because of a collision the station waits for a random amount
of time and sends it again. This waiting time must be random otherwise the same
frames will collide again and again.
• Therefore, pure ALOHA dictates that when the time-out period passes, each station
must wait for a random amount of time before resending its frame. This randomness
will help avoid more collisions.
• The pure ALOHA protocol has a second method to prevent congesting the channel
with retransmissions. It is shown in Fig. 2.16.
• A node or station sends the frame. Then wait for a period of time which is 2 times the
maximum propagation delay.
• If the acknowledgement is received, the transmission is successful. If the
acknowledgement is not given, then the station has to use a back-off strategy and send
the packet again. After several tries, if it is not happening then procedure is aborted.
• Vulnerable Time: Vulnerable time is a time in which there is possibility of collision. It
is calculated as,
Pure ALOHA vulnerable time = 2 × Tfr
2.25
Fig. 2.16: ALOHA Protocol Procedure
• Throughput: Let G be the average number of frames generated by the system during
one frame transmission time. Then the average number of successful transmissions
for pure ALOHA is S = G × e -2G. The maximum throughput of pure ALOHA is 0.184 i.e.
not more than 18.4 %.
Slotted ALOHA: (April 16; Oct. 17)
• An improvement to the original ALOHA protocol was "Slotted ALOHA", which
introduced discrete time slots and increased the maximum throughput.
• Slotted ALOHA was invented to improve the efficiency of pure ALOHA as chances of
collision in pure ALOHA are very high.
• In slotted ALOHA, the time of the shared channel is divided into discrete intervals
called slots.
• The stations can send a frame only at the beginning of the synchronized time slot and
only one frame is sent in each slot.
• In slotted ALOHA, if any station is not able to place the frame onto the channel at the
beginning of the slot, i.e. it misses the time slot then the station has to wait until the
beginning of the next time slot.
• In slotted ALOHA, there is still a possibility of collision if two stations try to send at the
beginning of the same time slot as shown in Fig. 2.17.
• Slotted ALOHA still has an edge over pure ALOHA as chances of collision are reduced
to one-half. Slotted ALOHA requires global time synchronization.
2.26
Collsion Collsion
Duration Duration
Frame 1.1 Frame 1.2
Time
Station 1
Frame 2.2
Frame 2.1
Time
Station 2
Frame 3.1 Frame 3.2
Time
Station 3
Frame 4.1 Frame 4.2

Time
Station 4 Slot 1 Slot 2 Slot 3 Slot 4 Slot 5 Slot 5
Fig. 2.17: Frames in Slotted ALOHA
2. Carrier Sense Multiple Access (CSMA): (April 17, 18; Oct. 17)
• To minimize the chance of collision and increase the performance, the CSMA method
was developed. The chance of a collision can be reduced if a station senses the
medium before trying to use it.
• Protocols in which stations listen for a carrier and act accordingly are called Carrier
Sense Protocols.
• CSMA is a network access method used on shared network topologies such as Ethernet
to control access to the network.
• Devices attached to the network cable listen (carrier sense) before transmitting. If the
channel is in use, devices wait before transmitting.
• MA (Multiple Access) indicates that many devices can connect to and share the same
network. All devices have equal access to use the network when it is clear.
• Even though devices attempt to sense whether the network is in use, there is a good
chance that two stations will attempt to access it at the same time.
• On large networks, the transmission time between one end of the cable and another is
enough that one station may access the cable even though another has already just
accessed it.
• The chances of collision still exist because of propagation delay. The frame transmitted
by one station takes some time to reach other stations.
• In the meantime, other stations may sense the channel to be idle and transmit their
frames. This results in the collision, as shown in Fig. 2.18.
2.27
2 Starts 3 Starts
at time t1 at time t2
1 2 3 4C
t1
t2
Area where
1’S signals exist
Area where
both signals exist
Area where
2’S signals exist
Time Time
Fig. 2.18: Time/Space Model of the Collision in CSMA
Vulnerable Time:
• The vulnerable time for CSMA is the propagation time Tp1, needed for a signal to
propagate from one end of the medium to the other.
• When a station sends a frame, and any other station tries to send, collision occurs. But
if the first bit of the frame reaches the end of the medium, other stations will already
have heard the bit and will stop ourselves from sending.
1 2 C3 4
2 Senses 3 Senses 4 Senses
here here here
t1
Vulnerable time
=
Propagation time
Frame Propagation
Time
Fig. 2.19: Vulnerable Time for CSMA

Persistence Methods:
• If the channel is free definitely stations will transmit data. But if the channel is busy,
what should a station do? Three different protocols are invented for this.
2.28
• There are three different types of CSMA protocols i.e., I-persistent CSMA, Non-
persistent CSMA and P-persistent CSMA as shown in Fig. 2.20.
CSMA
1-persistent Non-persistent P-persistent
Fig. 2.20: Types of CSMA
1-persistent CSMA:
• This method is very simple and straightforward. In 1-persistent CSMA method, a
station that wants to transmit data continuously senses the channel to check whether
the channel is idle or busy. If the channel is busy, the station waits until it becomes
idle.
• When the station detects an idle-channel, it immediately transmits the frame with
probability 1. Hence, it is called I-persistent CSMA.
• This method has the highest chance of collision because two or more stations may find
the channels to be idle at the same time and transmit their frames.
• When the collision occurs, the stations wait a random amount of time and start all
over again.
Drawback of 1-persistent CSMA:

• The propagation delay time greatly affects this protocol. Let us suppose, just after the
station I begins its transmission, station 2 also becomes ready to send its data and
senses the channel.
• If the station I signal has not yet reached station 2, station 2 will sense the channel to
be idle and will begin its transmission. This will result in a collision.
Sense and Transmit
Continuously sense
Channel ?
Time Busy
Idle
idle channel
Busy channel
Station can transmit
Fig. 2.21: 1-persistent CSMA
2.29
• Even if propagation delay time is zero, collision will still occur. If two stations become
ready in the middle of the third station's transmission, both stations will wait until the
transmission of the first station ends and then both will begin their transmission
exactly simultaneously. This will also result in collisions.
Non-persistent CSMA:
• In non-persistent CSMA, a station that has a frame to send senses to the channel. If the
channel is idle, it sends immediately.
• If the channel is busy, it waits for a random amount of time and then senses the
channel again.
• In non-persistent CSMA the station does not continuously sense the channel for the
purpose of capturing it when it detects the end of previous transmission.
Busy Wait
Channel ? Randomely
Idle
Station can transmit
Sense and Transmit
Sense Sense
Wait Wait
Time
Idle channel
Busy channel
Fig. 2.22: Non-persistent CSMA
Advantage of Non-persistent CSMA:

1. It reduces the chance of collision because the stations wait a random amount of
time.
2. It is unlikely that two or more stations will wait for the same amount of time and
will retransmit at the same time.
Disadvantage of Non-persistent CSMA:
1. It reduces the efficiency of the network because the channel remains idle when
there may be stations with frames to send.
2. This is due to the fact that the stations wait a random amount of time after the
collision.
2.30
P-persistent CSMA:
• This method is used when a channel has time slots such that the time slot duration is
equal to or greater than the maximum propagation delay time.
• Whenever a station becomes ready to send, it senses the channel. If the channel is
busy, the station waits until the next slot.
• If the channel is idle, it transmits with a probability p. With the probability q=l-p, the
station then waits for the beginning of the next time slot.
• If the next slot is also idle, it either transmits or waits again with probabilities p and q.
This process is repeated till either frame has been transmitted or another station has
begun transmitting.
• In case of the transmission by another station, the station acts as though a collision has
occurred and it waits a random amount of time and starts again.
Probability outcome does
Continuously sense not allow transmission
Transmit
Time slot Time slot Time slot Time slot
Time
Channel ?
Busy
Idle
Idle
>p Probability
Channel ? Wait a slot
outcome
Busy <=P
Acts as a though collision has Station can transmit

occurred and Start again
Fig. 2.23: P-persistent CSMA
• Advantage of p-persistent CSMA is it reduces the chance of collision and improves the
efficiency of the network.
3. CSMA/CD: (April 16; Oct. 17)
• CSMA/CD stands for Carrier Sense Multiple Access with Collision Detection. CSMA/CD is
a protocol in which the station senses the carrier or channel before transmitting
frames just as in persistent and non-persistent CSMA.
2.31
• If the channel is busy, the station waits. The wastage of channel time problem is
handled in CSMA/CD.
• In CSMA/CD, the station that places its data onto the channel after sensing the channel
continues to sense the channel even after the data transmission.
• If a collision is detected, the station aborts its transmission and waits for a
predetermined amount of time and then sends its data again. As soon as a collision is
detected, the transmitting station releases a jam signal.
• Jam signals will alert the other stations. The stations are not supposed to transmit
immediately after the collision has occurred. Otherwise there is a possibility that the
same frames would collide again.
• After some back-off delay time the stations will retry the transmission. If the collision
occurs again then the back off delay time is increased progressively.
• Therefore, the CSMA/CD method consists of alternating transmission periods and
collisions with idle periods when none of the stations is transmitting.
Collision
Idle period
Frame Frame
Time
Transmission Transmission
period period
Fig. 2.24: CSMA/CD with Three States
• The entire scheme of CSMA/CD is depicted in Figs. 2.25 and 2.26.
1 2 3 4
t1
First bit of 1
Transmission t2
Transmission
time
t3 time
f C
t4 First bit o 3’ s collision
1’ s collision detection
detection Collision and abortion
and abortion occurs
Time
Fig. 2.25: Collision of the First Bit in CSMA/CD
2.32
1 2 3 4
Collision
t1 occurs
Transmission t2 Transmission
time Part of 1’s frame t3 time
frame
t4 Part of 3’s
1 detects
collision and
aborts
3 detects
collision and
Time aborts Time
Fig. 2.26: Collision and Abortion in CSMA/CD
CSMA/CD Procedure:
• Fig. 2.27 shows a flow chart for the CSMA/CD protocol.
Note: Station has

Start
K : Number of attempts a frame to send
Tp : Maximum propagation time
Tfr : Average transmission time for a frame
K=0
TB : Back - off time
Apply one of the

persistence methods
(1- persistence, non-persistent,
or p-persistent)
Eligible for transmission
Transmission done
Wait TB time Yes
or
(TB =R X or R X Tfr) Collision detected
No
Choose a random
Transmit
number R between
K and Receive
0 and 2 - 1
No
Kmax is Send a Yes Collision
K > Kmax K=K+1
normally 15 jamming signal detected
Yes No
Abort Success
Fig. 2.27: Flowchart for the CSMA/CD Protocol
2.33
Explanation of Fig. 2.27:

• The station that has a ready frame sets the back off parameter to zero.
• Then it senses the line using one of the persistent strategies.
• If so, send the frame. If there is no collision for a period corresponding to one
complete frame, then the transmission is successful.
• Otherwise the station sends the jam signal to inform the other stations about the
collision.
• The station then increments the back off time and waits for a random back off time
and sends the frame again.
• If the back off has reached its limit then the station aborts the transmission.
• CSMA/CD is used for the traditional Ethernet.
• CSMA/CD is an important protocol. IEEE 802.3 (Ethernet) is an example of CSMA/CD. It
is an international standard.
• The MAC sublayer protocol does not guarantee reliable delivery. Even in absence of
collision the receiver may not have copied the frame correctly.
4. CSMA/CA: (Oct. 17, April 18)
• CSMA/CA stands for Carrier Sense Multiple Access with Collision Avoidance. The basic
idea behind CSMA/CD is that a station needs to be able to receive while transmitting to
detect a collision.
• When there is no collision, the station receives its own signal. When there is a
collision, the station receives two signals, its own and signals transmitted by another
station.
• Station needs to distinguish between these two signals. In wired networks if collision
occurs detected energy is almost double.
• Whereas in wireless networks much of the energy is lost in transmission only. So it
becomes difficult to detect collisions.
• It is particularly important for wireless networks, where the collision detection of the
alternative CSMA/CD is unreliable due to the hidden node problem.
• So to avoid collision in wireless networks, because they cannot be detected we need
another protocol. Carrier sense multiple access with collision avoidance (CSMA/CA)
was invented for this network.
• CSMA/CA avoids the collisions using three basic techniques as shown in the Fig. 2.28.
(i) Inter frame space,
(ii) Contention window, and
(iii) Acknowledgements.
2.34
Found idle Size:Binary

eponential
IFS
Time
Contention window
Busy Send frame Time out
Fig. 2.28: Three Basic Techniques for Collision Avoidance
1. Inter-Frame Space (IFS):

• Whenever the channel is found idle, the station does not transmit immediately. It
waits for a period of time called Inter Frame Space (IFS).
• When the channel is sensed to be idle, it may be possible that the same distant station
may have already started transmitting and the signal of that distant station has not yet
reached other stations.
• Therefore, the purpose of IFS time is to allow this transmitted signal to reach other
stations.
• If after this IFS time, the channel is still idle, the station can send, but it still needs to
wait a time equal to contention time.
• IFS variables can also be used to define the priority of a station or a frame.
2. Contention Window:
• The Contention window is an amount of time divided into slots. A station that is ready
to send chooses a random number of slots as its wait time.
• The number of slots in the window changes according to the binary exponential back-
off strategy. It means that it is set for one slot the first time and then doubles each time
the station cannot detect an idle channel after the IFS time.
• This is very similar to the p-persistent method except that a random outcome defines
the number of slots taken by the waiting station.
• In the contention window the station needs to sense the channel after each time slot.
• If the station finds the channel busy, it does not restart the process. It just stops the
timer and restarts it when the channel is sensed as idle.
3. Acknowledgement:
• Despite all the precautions, collisions may occur and destroy the data.
• The positive acknowledgment and the time-out timer can help guarantee that the
receiver has received the frame.
CSMA/CA Procedure:
• Fig. 2.29 shows the flow chart explaining the principle of CSMA/CA.
2.35
Fig. 2.29: CSMA/CA Procedure

Explanation of Fig. 2.29:
• This is the CSMA protocol with collision avoidance.
• The station ready to transmit, senses the line by using one of the persistent strategies.
• As soon as it finds the line to be idle, the station waits for an IFG (Interframe gap)
amount of time.
• If so waits for some random time and sends the frame.
• After sending the frame, it sets a timer and waits for the acknowledgement from the
receiver.
• If the acknowledgement is received before expiry of the timer, then the transmission is
successful.
• But if the transmitting station does not receive the expected acknowledgement before
the timer expires then it increments the back off parameter, waits for the back off time
and re-senses the line.
2.36
2.4.4 Controlled Access

• In controlled access, the stations consult each other to find which station has the right
to send. A station cannot send unless it is authorized by other stations.
• Controlled access protocols grant permission to send data only to one node at a time,
avoiding collision on the shared medium.
• There are three methods in the controlled access area namely, Reservation, Polling,
and Token passing.
1. Reservation: (Oct. 18; April 19)
• In this method, the station needs to make a "before sending data". The time is divided
into intervals. If there are m stations, then the intervals are exactly m.
• Each interval will belong to a station. In its own slot also, it has to make reservations to
send the data frame. In each interval a reservation frame precedes the data frame.
• Fig. 2.30 shows this scenario.
Fig. 2.30: Reservation Method in the Controlled Access
• Fig. 2.30 shows five stations. In the first, reservation frame stations 1, 3 and 4 have
made the reservations.
• So the reservation frame is first followed by data frames of station 1, 3 and 4. In the
second interval, only station 1 has made the reservation.
2. Polling: (April 16, 19)
• In this method, one station is designated as a primary station and others are secondary
stations. All data transfer should be made through the primary device.
• Primary device controls the link and acts as a master. All secondary devices follow its
instructions.
• The primary device decides as to which device is allowed to use the channel. If the
primary station wants to receive data, it asks the secondaries if they have anything to
send. This method is called polling.
• If the primary station wants to send data, it asks the secondary station to receive the
data. This is called selecting.
1. Select: (April 16, 17, 18, 19; Oct. 17, 18)
• This is used when the primary device sends the data. When the primary station wants
to send data to other machines, it should intimate the other secondary device that it is
sending the data.
2.37
• To do this, the primary machine first sends the SEL frame in which the address of the
machine (secondary machine) to which it wants to send the data is present.
• Then, the primary machine waits for the acknowledgement from that machine. Once
the acknowledgement is reached then the data is sent.
2. Poll:
• Whenever primary wants to receive data then polling is used. It polls each machine, if
it has something to send.
• The secondary response with MAK frame which means nothing to send or with data
frame if it wants to send data.
• If the response from the secondary device is negative i.e. received MAK frame, then
the next machine is polled until it finds one with data to send.
• When the primary machine gets the data, it accepts the data frame and returns the
acknowledgement.
• Fig. 2.31 shows select and poll functions in the polling access method.
A B C A B C
Primary Primary
SEL Poll
ACK
NAK
Select
Data Poll
Poll
ACK
Data
ACK
(a) (b)
(a) Select Function (b) Poll Function
Fig. 2.31
3. Token Passing:
• In this method, a station is allowed to send data when the station receives a special
frame called a token.
• In token passing, the stations are organized in a logical ring. For every station there is
a predecessor and a successor.
• The right to access the channel has been passed from predecessor to the current
station. The right will be passed to the successor when the current station has no more
data to send.
• Fig. 2.32 shows concepts of token passing.
2.38
A B C D
Fig. 2.32: Token Passing
• When the channel is idle then the token circulates around the ring. When the station
wants to send data, it waits for a token, when the token comes, the station captures the
token and sends the data. When the machine finishes the data, it releases the token to
the next machine (the successor).
• Token management is needed for this access method. The token must be monitored to
ensure it has not been lost or destroyed.
• The procedure is as shown in Fig. 2.33.
Fig. 2.33: Flowchart for Token Passing Procedure
2.4.5 Channelization (April 16, 17, 19; Oct. 18)

• Channelization is the multiple access method in which the available bandwidth of a
link is shared in time, frequency or through code between different stations.
• Sometimes channelization is also called as channel partition. The three channelization
protocols present in networking are FDMA, TDMA and CDMA.
• In FDMA (Frequency Division Multiple Access), the bandwidth is shared by all stations.
Each band is given to the station.
• Station sends data in its allocated band. The band belongs to the station all the time. It
is data link layer protocol. Here, the bandwidth is divided into channels.
2.39
• In TDMA (Time Division Multiple Access), the entire bandwidth is one channel. Each
station is allocated a time slice. During its time slice, the data is sent.
• In CDMA (Code Division Multiple Access), only one channel occupies the entire
bandwidth of the link. All stations can send data simultaneously.
• Here, the channel carries all transmissions simultaneously. Each station is assigned a
code which is a sequence number called chip.
1. FDMA: (April 17, Oct. 18)
• FDMA is a channel access method used in multiple-access protocols as channelization
protocol.
• In FDMA, the available bandwidth is divided into various frequency bands. Each
station is allocated a band to send its data. This band is reserved for that station for all
the time.
• The frequency bands of different stations are separated by small bands of unused
frequency. These unused frequency bands are called guard bands that prevent station
interference.
• FDM is a physical layer technique whereas FDMA is an access method in the data link
layer.
• FDMA is the process of dividing one channel or bandwidth into multiple individual
bands, each for use by a single user.
• Each individual band or channel is wide enough to accommodate the signal spectra of
the transmissions to be propagated.
Data Data
f f
A B
t t
Common
Channel
Data
f f
C D
t t
Silent Data
Fig. 2.34: Concept of FDMA
2.40
• The data to be transmitted is modulated on to each subcarrier, and all of them are
linearly mixed together.
• The best example of FDMA is the cable television system. Fig. 2.34 shows the FDMA
concept.
2. TDMA:
• TDMA is a channel access method for stored medium networks.
• TDMA stations share the bandwidth of the channel in time. Every station is allocated a
time slot, in which it can send data.
• TDMA is a digital technique that divides a single channel or band into time slots.
• In TDMA every station needs to know the beginning of its slot and the location of its
slot. To compensate for the delays guard spaces are used.
• In TDMA, the bandwidth is just one channel that is time shared between different
stations.
Data Data
f f
A B
t t
Common
Channel
Data
f f
C D
t t
Silent Data
Fig. 2.35: TDMA Method
• TDMA and TDM seem to be conceptually the same, but they are different. TDM is a
physical layer technique, whereas TDMA is a Data link layer access method.
• The Data link layer in each station tells its Physical layer to use the allocated time slot.
3. CDMA: (Oct. 17)
• CDMA means communication with different codes. For example, consider a large hall
with many people. Some of them understand only English, some Hindi, some Chinese
etc.
2.41
• The people communicate with a language, which they understand. The common
channel (hall) allows communication between several people, but in different
languages (codes).
• CDMA is also called spread-spectrum and code division multiplexing, one of the
competing transmission technologies for digital mobile phones.
• The transmitter mixes the packets constituting a message into the digital signal stream
in an order determined by a Pseudo-Random Number (PRN) sequence.
• PRN is also known to the intended receiver, which uses it to extract those parts of the
signal intended for itself. Hence, each different random sequence corresponds to a
separate communication channel.
• Unlike TDMA, in CDMA all stations can transmit data simultaneously, there is no
timesharing. CDMA allows each station to transmit over the entire frequency spectrum
all the time.
• Multiple simultaneous transmissions are separated using coding theory. In CDMA each
user is given a unique code sequence.
2.5 SWITCHING AND TCP/IP LAYERS

• Switching can happen at several layers of the TCP/IP protocol suite like Physical Layer,
Data Link Layer and Application Layer.
• At the physical layer of the TCP/IP protocol suite, we can have only circuit switching.
• There are no packets exchanged at the physical layer. The switches at the physical
layer allow signals to travel in one path or another.
• At the data link layer of the TCP/IP protocol suite, we can have packet switching.
However, the term packet in this case means frames or cells.
• Packet switching at the data link layer of the TCP/IP protocol suite is normally done
using a virtual-circuit approach.
• At the application layer of the TCP/IP protocol suite, we can have only message
switching. The communication at the application layer occurs by exchanging
messages.
• Conceptually, we can say that communication using e-mail is a kind of message-
switched communication, but we do not see any network that actually can be called a
message-switched network.
2.5.1 Types of Switching

• A network is a set of connected multiple devices, whenever multiple devices are there,
we have the problem of how to connect them to make one-to-one communication
possible.
2.42
• If a network is a LAN, we can have point to point or broadcast links. But for WAN,
such topologies are not possible.
• A better solution is switching. A switched network consists of a series of interlinked
nodes, called switches.
• Switches are devices capable of creating temporary connection between two or more
devices linked to the switch.
• In switched networks, some of these nodes are connected to the end systems
(computers or telephones). Others are used only for routing.
• Fig. 2.36 shows switched network.
Fig. 2.36: Switched Network

• There are basically three types of switching methods available namely, Circuit
Switching, Packet Switching and Message Switching as shown in Fig. 2.37.
Fig. 2.37: Types of Switching

• Circuit and packet switching are commonly used today. Message switching has been
phased out in general communications.
2.5.1.1 Circuit Switching (April 16, 18)
• Circuit switching takes place at the physical layer of the TCP/IP reference model.
Circuit switched networks consist of a set of switches connected by physical links.
2.43
• In a circuit switched network, two nodes communicate with each other over a
dedicated communication path.
• Circuit switching is commonly used technique in telephony, where the caller sends a
special message with the address of the callee (i.e. by dialing a number) to state its
destination.
• Fig. 2.38 shows the concept of circuit switching. In Fig. 2.38, six different rectangles
are shown. Each rectangle represents a carrier switching office (end office, toll office
etc.).
• As an example, we have shown every office has three incoming and three outgoing
lines.
• When a call passes through a switching office, a physical conceptual temporary
connection is established between the line on which the call came in and one of the
output lines, as shown by dotted lines.
• Once a call has been set up, a dedicated path between both ends exists and will
continue to exist until the call is finished.
Fig. 2.38: Circuit Switching
• Communication via circuit switching implies that there is a dedicated communication

path between the two stations.
• The path is connected through a sequence of links between network nodes. It involves
the following three distinct steps:
1. Circuit Establishment: To establish an end-to-end connection before any transfer
of data. Some segments of the circuit may be a dedicated link, while some other
segments may be shared.
2. Data Transfer: Transfer data is from the source to the destination. The data
may be analog or digital, depending on the nature of the network. The connection
is generally full-duplex.
2.44
3. Circuit Disconnect: Terminate connection at the end of data transfer. Signals

must be propagated to deallocate the dedicated resources
• Thus, the actual physical electrical path or circuit between the source and destination
host must be established before the message is transmitted.
• This connection, once established, remains exclusive and continuous for the complete
duration of information exchange and the circuit becomes disconnected only when
the source wants to do so.
Advantages of Circuit Switching:
1. The dedicated path/circuit established between sender and receiver provides a
guaranteed data rate.
2. Once the circuit is established, data is transmitted without any delay as there is no
waiting time at each switch.
3. Less expensive.
4. Fixed transit delays and throughput.
5. No loss of packets or out of order packets here as this is connection oriented
network unlike packet switched network.
Disadvantages of Circuit Switching:
1. As is it designed for voice traffic, it is not suitable for data transmission.
2. Circuit switching is usually done using a fixed rate stream (for example, 64 Kbps).
3. Circuit switched networks do not provide flow control nor error control.
4. Dedicated channels require more bandwidth.
5. It is more expensive compared to other techniques due to dedicated path
requirements.
6. Because connections are not shared, more links must be available to ensure new
connections are not blocked.
2.5.1.2 Packet Switching (Oct. 17)

• In packet switching, messages are divided into packets of fixed or variable size. The
size of the packet is determined by the network and the governing protocol.
• Packet switching approach was developed for long-distance data communication
(1970) and it has evolved over time.
• In packet switching is shown in Fig. 2.39. The data are transmitted in short packets
(few Kbytes).
• In packet switching, each packet has source and destination addresses, travelling from
one point (router) to the other point (router).
• Every packet contains some control information in its header, which is required for
routing and other purposes.
2.45
Fig. 2.39: Packet Switching
Advantages:
1. In packet switching, no circuit set up is required in advance.
2. In packet switching, the quality of data transmission is kept high (error free).
3. In packet switching, no bandwidth is reserved.
4. Packet switching is more fault tolerant.
5. Better utilization of the network segments in terms of the usage of the network
path.
6. Efficient for busty data and efficient used network paths.
7. In packet switching, computers at each node allow dynamic data routing.
Disadvantages:
1. In packet switching, different packets can follow different paths, so they may
arrive out of order.
2. If the network becomes overloaded, packets are delayed or discarded or dropped.
This leads to retransmission of lost packets by the sender. This often leads to loss
of critical information if errors are not recovered.
3. Packet switching network cannot be used in applications requiring very little delay
and higher quality of service e.g., reliable voice calls.
4. Protocols used in the packet switching are complex and require high initial
implementation costs.
• The two types of packet switching are Datagram Packet Switching and Virtual Circuit
Packet Switching.
Datagram Packet Switching: (Oct. 18)
• In a datagram network, each packet is routed individually by network devices based
on the destination address contained within each packet.
• Due to each packet is routed individually, the result is that each packet is delivered
out-of-order with different paths of transmission, it depend on the networking devices
like (switches and routers) at any given time.
2.46
• After reaching the recipient location, the packets are reassembling to the original
form. Datagram packet switching is normally implemented in the network layer. Each
packet carries a header that contains the full information about the destination.
• When the switch receives the packet, the destination address in the header of the
packet is examined; the routing table is consulted to find the corresponding port
through which the packet should be forwarded.
Fig. 2.40: Datagram Packet Switching

Advantages of Datagram Packet Switching:
1. No call setup phase required.
2. More flexible because routing can be used to avoid congested ports of the network.
3. Cheaper in cost.
Disadvantages Datagram Packet Switching:
1. Packets are forwarded slowly as compared to the virtual circuit approach.
2. Processing time requests more at node.
Virtual Circuit Packet Switching: (Oct. 18)
• In this type of network switching, packets are sent in sequential order over a defined
route. Virtual circuit packet switching is normally done at the data link layer.
• Virtual circuit packet switching establishes a fixed path between a source and a
destination to transfer the packets.
• A source and destination have to go through three phases in a virtual circuit packet
switching:
1. Setup phase.
2.47
2. Data transfer phase.

3. Connection release phase.
• A logical connection is established when a sender sends a setup request to the receiver
and the receiver sends back an acknowledgement to the sender if the receiver agrees.
• All packets belonging to the same source and destination travel the same path. The
information is delivered to the receiver in the same order as transmitted by the
sender.
Fig. 2.41: Virtual Circuit Packet Switching

Advantages of Virtual Packet Circuit Switching:
1. Virtual circuit provides packet sequencing and error control.
2. Packet forwarding is fast and quick.
3. Multiple packets sent by the same source to the same destination.
Disadvantages of Virtual Packet Circuit Switching:
1. Loss of a node losses all circuits through that node so it is less reliable.
2. Less flexible than other approaches.
3. Cost is higher than the datagram approach.
Virtual Circuit Approaches:
1. Switched Virtual Circuit (SVC): SVC format is comparable to dial up lines in
circuit switching.
2. Permanent Virtual Circuit (PVC): These are comparable to leased line in circuit
switching.
2.48
Comparison between SVC and PVC:

Sr.
SVC PVC
No.
1. SVC stands for Switched Virtual PVC stands for Permanent Virtual
Circuit. Circuit.
2. In SVC a source and a destination In PVC a source and a destination may
connect when data is being choose to have a PVC.
transferred.
3. SVC creates a temporary short PVC creates a permanent and continuous
connection. connection.
4. Cheapest in cost. Cost is high.
5. Examples: ATM, X.25, etc. Examples: Frame Relay, X.25, etc.
Comparison of Datagram Approach and Virtual Circuit Packet Switching:

Sr. Datagram Packet Virtual Circuit Packet
No. Switching Switching
1. All packets are free to go to any path First packet goes and reserves resources
on any intermediate router which is for the subsequent packets which as a
decided on the go by dynamically result follow the same path for the whole
changing routing tables on routers. connection time.
2. More flexible because routing can be Less flexible.
used to avoid congested port of the
network.
3. Slow in packet forwarding. Packets are forwarded quickly.
4. More reliable. Less reliable because loss of node losses
all circuits through that node.
5. It is a connectionless service. It is a connection-oriented service.
6. No circuits or path established. Virtual circuits are established between
source and destination before data
transfer.
7. Out of order data delivery. Sequential type of delivery.
8. No resource allocation required. Resources are allocated on demand.
Contd...
2.49
9. Unreliable. Virtual circuits are highly Reliable.

10. Unpredictable delay. High delay.
11. Example: Internet. Example: X.25, Frame Relay.
12. Implementation is easy and cost Implementation of virtual circuits is
efficient. costly.
2.5.1.3 Message Switching

• In message switching, it is not necessary to establish a dedicated path between
transmitter and receiver. In this, each message is routed independently through the
network.
• In message switching, each message carries a header that contains the full information
about the destination.
• In message switching, the message is sent to the nearest directly connected switching
node as shown in Fig. 2.42.
• This node stores the message, checks for errors, selects the best available route and
forwards the message to the next intermediate node.
• Each intermediate device receives the message and stores it until the nest device is
ready to receive it and then this message is forwarded to the next device.
• For this reason, a message switching network is sometimes called a Store and
Forward Switching.
C E
M
es
sa
ge
A B 2 C H
Message 1
D F
Path of Message 2
Path of Message 1
Fig. 2.42: Message Switching Technique
Advantages:
1. In message switching, no circuit setup is required in advance.
2. As more devices share the same channel simultaneously for message transfer, it
has higher channel efficiency compared to circuit switching.
3. Data rate conversion is possible in message switching.
2.50
4. Message switching, supports message lengths of unlimited size.

5. It is possible to incorporate priorities to the messages as they use store and
forward technique for delivery.
Disadvantages:
1. Message switching type does not establish a dedicated path between the devices.
As there is no direct link between sender and receiver, it is not reliable
communication.
2. The method is costly as store and forward devices are expensive. This is due to
large storage disks requirements to store long messages for long duration.
3. Message switching is not compatible for interactive applications such as voice and
video. This is due to longer message delivery time.
4. Message switching is very slow because of store-and-forward technique.
Difference between Message Switching, Packet Switching and Circuit Switching:
Sr. Message Circuit Packet

Parameters
No. Switching Switching Switching
1. Concept In message switching, When the computer With this
each switch stores the places a telephone technology,
whole message and call, the switching packets are sent
forwards it to the next equipment within as soon as they
switch. Although we the telephone system are available.
don't see message seeks out a physical
switching at lower path all the way
layers, it is still used in from your telephone
some applications like to the receiver’s
electronic mail (e- telephone. This
mail). technique is called
circuit switching.
2. Store and Yes. No. Yes.
forward
transmission
3. Addressing Geographical Hierarchical Hierarchical
addresses. numbering plan. address space.
4. Routing Manual routing. Route selected Each packet
during call setup. routed
independently.
Contd...
2.51
5. Multiplexing Character Circuit multiplexing. Packet

multiplexing, message multiplexing
multiplexing. shared media
across
networks.
6. Call setup No. Required. Not needed.
7. Dedicated Not required. Yes. No.
physical path
8. Bandwidth -- Fixed. Dynamic.
available
9. Application Telegraph network for Telephone network Internet for
transmission of for bidirectional, datagram and
telegrams. real time transfer of reliable stream
voice signals. service between
computers.
10. End Terminal Telegraph, Teletype. Telephone, modem. Computer.
11. Information Morse, Baudot, ASCII. Analog voice or PCM Binary
Type digital voice. Information.
12. Transmission Digital data over Analog and Digital Digital data over
system different transmission data over different differential
media. transmission media. transmission
media.
2.6 WIRED LANs

• In Chapter 1, we learned that a Local Area Network (LAN) is a computer network for
the home and small business is designed for a limited geographic area (less than) such
as a building or campus.
• Most LANs today are also linked to a Wide Area Network (WAN) or the Internet.
• Wired networks are the most common type of Local Area Network (LAN) technology.
A wired network (LAN) is simply a collection of two or more computers, printers, and
other devices linked by Ethernet cables.
• LANs can be built using either wired or wireless technology. LAN supports several
technologies such as Ethernet, Token Ring, Token Bus, FDDI, and ATM LAN.
• The Institute of Electrical and Electronic Engineers (IEEE) developed a series of
networking standards for LANs. These standards are collectively known as IEEE 802.
2.52
• Today Ethernet is by far the dominant technology. Other technologies survived for a
while. Fig. 2.43 shows an example of four computers connected in a traditional
Ethernet LAN.
Fig. 2.43: A Traditional Ethernet LAN

• The IEEE has subdivided the data link layer into two sub layers i.e. LLC (Logical Link
Control) and MAC (Media Access Control).
Advantages to using a Wired LAN:
1. Speed: Wired LANs almost invariably provide higher speed connections (basically
due to the increased reliability).
2. Reliability: Wired LANs tend to be much more reliable as they have a dedicated
wire, suitably insulated, down which the router or server can converse with any
clients.
3. Better Security: It's more difficult to tap into a (suitably secured) wired LANs than
a wireless one as there is no airborne signal that can be picked up.
4. Energy Saving: Less electricity is consumed if your router can send a stream of
electrons down a wire instead of having to broadcast on the airwaves.
Disadvantages of Wired LAN:
1. Difficult to Physical Setup: Wired LANs require cables to be installed between
your router, hub, switch, printer and computers location(s).
2. Security Problems: Wired LANs may well require more stringent security setup
in order to secure them properly.
2.53
3. Mobility: Wired LANs do not provide wireless mobility.

4. Cost: High cost because of the cost of cables, hub, connectors etc.
5. Time Consuming: Ethernet cables must be run from each computer to another
computer or to the central device like hub or switch. It can be time-consuming and
difficult to run cables under the floor or through walls, especially when computers
sit in different rooms.
2.6.1 Standard Ethernet (April 16, 18; Oct. 18)

• The original Ethernet was developed in 1976 at Xerox’s Palo Alto Research center
(PARC). The original Ethernet technology with the data rate of 10 Mbps refer to the
standard Ethernet.
• Ethernet is a standardized system for connecting computers to a Local Area Network
(LAN). The Institute for Electrical and Electronic Engineers (IEEE) developed an
Ethernet standard known as IEEE Standard 802.3.
• The Standard 802.3 defines rules for configuring an Ethernet network and also
specifies how the elements in an Ethernet network interact with one another.
• The 802.3 standard defines rules for configuring an Ethernet network and also
specifies how the elements in an Ethernet network interact with one another.
• Since then, it has gone through four generations i.e., Standard Ethernet (10 Mbps), Fast
Ethernet (100 Mbps), Gigabit Ethernet (1 Gbps) and Ten-Gigabit Ethernet (10 Gbps) as
shown in Fig. 2.44.
Fig. 2.44: Four generations of Ethernet evolution
2.6.1.1 Characteristics
• Some characteristics of the Standard Ethernet are explained below:
1. Connectionless and Unreliable Service:
• Ethernet provides a connectionless service which means each frame sent is
independent of the previous or next frame.
• Ethernet has no connection establishment or connection termination phases. The
sender sends a frame whenever it has it; the receiver may or may not be ready for it.
2.54
• The sender may overwhelm the receiver with frames, which may result in dropping
frames. If a frame drops, the sender will not know about it.
• Since IP, which is using the service of Ethernet, is also connectionless, it will not know
about it either. If the transport layer is also a connectionless protocol, such as UDP, the
frame is lost and salvation may only come from the application layer.
• However, if the transport layer is TCP, the sender TCP does not receive
acknowledgement for its segment and sends it again.
2. Unreliable Service:
• Ethernet is also unreliable like IP (Internet Protocol) and UDP (User Datagram
Protocol).
• If a frame is corrupted during data transmission and the receiver finds out about the
corruption, which has a high level of probability of happening because of the CRC-32,
the receiver drops the frame silently. It is the duty of high-level protocols to find out
about it.
3. Frame Format: (April 18; Oct., 18)
• The format of the Ethernet MAC frame is shown in Fig. 2.45, and it contains several
fields.
• Ethernet does not provide any mechanism for acknowledgment of received frames.
Acknowledgments must be implemented at the higher layers.
Preamble : 56 bits of alternating 1s and 0s.

SFD : Start frame delimiter, flag (10101011)
Destination Source Length

Preamble SFD Data and padding CRC
address address or type
7 bytes 1 byte 6 bytes 6 bytes 2 bytes 4 bytes
Physical layer
header
Fig. 2.45: 802.3 Ethernet MAC Frame
• The fields in the frame format are:

(i) Preamble: This is the first field of the 802.3 frame, which contains 7 bytes (56
bits) of alternating 0s and 1s. Which indicates the receiving system about the
coming frame and enables it to synchronize its input timing. The preamble is
actually added at the physical layer.
(ii) Start Frame Delimiter (SFD): SFD is the second field in the frame. Size of this
field is 1 byte (1 byte : 10101011). This field signals the beginning of the frame.
2.55
The SFD warns the station or stations that this is the last chance for
synchronization. The last 2 bits is 11 and alerts the receiver that the next field is
the destination address.
(iii) Destination Address (DA): The DA field is of 6 bytes and contains the physical
address of the destination station or stations to receive the packet.
(iv) Source Address (SA): The source address field is also 6 bytes and contains the
physical address of the sender of the packet.
(v) Length or type: This field is defined as a type field or length field. Both uses are
common. The original Ethernet used this field as the type field to define the
upper-layer protocol using the MAC frame. The IEEE standard used it as the
length field to define the number of bytes in the data field.
(vi) Data: This field contains data encapsulated from the upper-layer protocols. Size
of data is minimum of 46 and a maximum of 1500 bytes.
(vii) CRC: This last field contains error detection information.
4. Frame Length:
• Ethernet has forced restrictions about both the minimum and maximum lengths of a
frame, as shown in Fig. 2.46.
Minimum payload length : 46 bytes
Maximum payload length : 1500 bytes
Destination Source Length Data and padding CRC

address address PDU
6 bytes 6 bytes 2 bytes 4 bytes
Minimum frame length : 512 bits or 64 bytes
Maximum frame length : 12,144 bits or 1518 bytes
Fig. 2.46: Minimum and Maximum of Ethernet Frame
• As we know CSMA/CD is used as an access protocol in Ethernet.

• The minimum length restriction is required for the correct operation of CSMA/CD.
Minimum length of the Ethernet frame is 512 bits or 64 bytes.
• Frame contains 6 bytes of source address, 6 bytes of destination address, 2 bytes of
length or type, and 4 bytes of CRC, total 18 bytes of header and trailer are required.
• The minimum length of data from the upper layer is 64-18=46 bytes. If the upper layer
packet is less than this, padding (adding extra bits) is done.
• The maximum length of an Ethernet frame without preamble and SFD field defined by
standard is 1518 bytes.
2.56
• If we subtract the 18 bytes of header and trailer, the maximum length of the payload is
1500 bytes.
• The maximum length restriction has following two reasons:
(i) Memory was very expensive when Ethernet was designed, this restriction helped
to reduce the size of the buffer.
(ii) The maximum length restriction prevents one station from taking the complete
control of the shared medium, blocking other stations that have data to send.
2.6.2 Addressing
• Every station on an Ethernet network like a PC, workstation, printer etc., has its own
Network Interface Card (NIC), installed inside the station.
• The NIC provides the station with a 6-byte (48 bit) physical address. Fig. 2.47 shows
physical address in hexadecimal notation.
06 : 01 : 02 : 01 : 2C : 4B
6 bytes = 12 hex digits = 48 bits
Fig. 2.47: Physical Address in Hexadecimal Notation
Unicast, Multicast and Broadcast Addresses:

• As we know a source address is always a unicast address. The frame comes from only
one station. But the destination address can be unicast, multicast or broadcast.
• Fig. 2.48 shows the difference between a unicast address and a multicast address.
• If the least significant bit of the first byte in a destination address is 0, the address is
unicast; otherwise, it is multicast.
Unicast : 0; multicast: 1
Byte 1 Byte 2 Byte 6
Fig. 2.48: The difference between Unicast Address a Multicast Address
• The broadcast address is a special case of the multicast address i.e. the recipients are
all the stations on the LAN. A broadcast destination address contains all forty-eight
(48) bits are 1s.
2.6.3 Access Method: CSMA/CD (April 18)
• Standard Ethernet uses 1-persistent CSMA/CD (Carrier Sense Multiple Access/Collision

Detection). 1-persistent mode waits for the medium to be idle, then transmits data.
2.57
• CSMA/CD is a type of contention protocol that defines how to respond when a collision
is detected, or when two devices attempt to transmit packages simultaneously.
• In an Ethernet network, slot time is defined in bits. It is the time required for a station
to send 512 bits. For example, for traditional 10mbps LAN it is 51.2µs and calculated as
follows
Slot time = Round-trip Time + Time required to send the Jam Sequence
Slot Time and Collision:
• A 512 bits slot time allows proper functioning of CSMA/CD. To understand this let us
consider two cases.
Case 1:
• In the first case, the sender sends a minimum size packet of 512 bits. Before the sender
can send the entire packet out, the signal travels through the network and reaches the
end of the network.
• If there is another signal at the end of the network collision occurs. The sender can
send a jam signal to abort the sending of a frame.
• The round trip time plus the time required to send the jam signal should be less than
the time needed for the sender to send the minimum frame, 512 bits.
Case 2:
• The sender must be aware of the collision before it has sent the entire frame.
• In the second case, if the sender sends a frame larger than the minimum size, after
sending the first 512 bits, it is guaranteed that collision will not occur during the
transmission of this frame.
• The entire medium belongs to the sender. The sender needs to listen for a collision
only during the time the first 512 bits are sent.
Slot Time and Maximum Network Length:
• A relationship between the slot time and maximum length of the network is
dependent on the propagation speed of the signal in the particular medium.
Max Length = Propagation Speed × Slot Time/2
2.6.4 Implementations (April 19)
• The four common Standard Ethernet implementations during the 1980s are shown in
Fig. 2.49.
• In Standard Ethernet implementation (10 Mbps) sender converts data into a digital
signal by using the Manchester scheme, at the receiver, received signals are
interpreted as Manchester and decoded into data.
2.58
Fig. 2.49: Categories of Standard Ethernet
• Fig. 2.50 shows the encoding scheme for Standard Ethernet.
Fig. 2.50: The encoding scheme for Standard Ethernet
1. 10Base5 (Thick Ethernet): (April 19)

• The first Ethernet implementation is called 10Base5, thick Ethernet, or Thicknet.
10Base5 was the first Ethernet specification used in bus topology.
• In 10Base5 specification external transceivers (transmitter and receiver) connected via
a tap of a thick coaxial cable.
• Fig. 2.51 shows 10base5 implementation.
PC1 PC2 PC3

10Base5
Server
10 Mbps 500 m Transceiver cable
maximum 50 m
Baseband Cable Cable
(digital) end end
Transceiver Thick coaxial cable
maximum 500 m
Fig. 2.51: 10Base5 Implementation
2.59
• The transceiver transmits, receives and detects collision. The transceiver is connected
to a station via transceiver cable which provides a separate path for sending and
receiving. Collision can occur only in the coaxial cable.
• In 10Base5 implementation, the maximum length of coaxial cable should not exceed
500 m, otherwise, signal quality is degraded. If cable length is required more than 500
m, a connecting device repeater is used.
2. 10Base2 (Thin Ethernet): (April 19)
• The second implementation is called 10Base2, thin Ethernet, or Cheapernet. The
10Base2 also uses a bus topology, it is more cost effective than 10Base5.
• The cable used in this implementation is much thinner, more flexible and can be bent.
Due to this cable can pass very close to the stations.
• The transceivers are built inside the Network Interface Card (NIC), which is installed
inside the station.
• Thin coaxial is cheaper than thick and the tee (T) connectors are also cheaper than
taps used in 10Base5.
• Installation of 10Base2 is simpler because the thin coaxial cable is very flexible. The
length of each segment cannot exceed 185 m (close to 200 m) due to the high level of
attenuation in thin coaxial cable.
• Fig. 2.52 shows the schematic diagram of a 10Base2 implementation.
Cable end
PC2
PC1
Server
10Base2
10 Mbps 185 m
Baseband Thin coaxial cable,

(digital) maximum 185 m
Tee
connectors
Cable end
Fig 2.52: 10Base2 Implementation
3. 10Base-T (Twisted-Pair Ethernet): (April 19)
• The third implementation is 10Base-T or twisted-pair Ethernet. 10Base-T uses physical
star topology.
• All stations are connected to a hub or switch via two pairs of twisted pair. From these
two pairs one pair is used for sending and other is used for receiving data in between
the station and the hub.
• To minimize the effect of attenuation in the twisted pair, its length is defined as 100 m
only.
2.60
• Fig. 2.53 shows 10Base-T Ethernet implementation.

PC2
Server
10Base-T
10 Mbps Twisted pair PC1
Baseband
(digital)
Two pairs of
UTP cable
10Base-T hub
Fig. 2.53: 10Base-T Implementation
4. 10Base-F (Fiber Ethernet): (April 19)
• 10Base-F uses a star topology to connect stations to a hub.
• The stations are connected to the hub using two fiber-optic cables as shown in the
Fig. 2.54.
PC1 PC2
Server
10Base-F
Two fiber-optic
10 Mbps Fiber cables
Baseband
(digital)
10Base-F hub
Fig. 2.54: 10Base-T Implementation
Summary of Standard Ethernet Implementation:
Sr.
Characteristics 10Base5 10Base2 10Base-T 10Base-F
No.
1. Media Thick coaxial Thin coaxial 2 UTP 2 Fiber
cable cable
2. Maximum length 500 m 185 m 100 m 2000 m

3. Line encoding Manchester Manchester Manchester Manchester
2.61
2.6.5 Fast Ethernet (April 15, 16)
• Fast Ethernet is an Ethernet standard for 100-Mbps data transmission defined by the
IEEE 802.3u specification.
• Fast Ethernet was designed to compete with LAN protocols such as FDDI or Fiber
Channel.
• Fast Ethernet is 10 times faster than standard Ethernet and operates at 100 Mbps. Fast
Ethernet is backward compatible to standard Ethernet.
Goals of Fast Ethernet: (April 16)
1. Upgrade the data rate to 100 Mbps.
2. Keep the same 48-bit address.
3. Make it compatible with Standard Ethernet.
4. Keep the same frame format.
5. Keep the same minimum and maximum frame length.
• Autonegotiation is a new feature added to Fast Ethernet. Autonegotiation allows a
station or a hub a range of capabilities. It allows two devices to negotiate the mode or
data rate of operation. CSMA/CD is an access method used by Fast Ethernet.
• Fig. 2.55 shows common Fast Ethernet implementation techniques.
• Fast Ethernet implementation at the physical layer can be categorized as either two-
wire or four-wire.
1. The two-wire implementation can be either category 5 UTP (100Base-TX) or fiber-
optic cable (100Base-FX).
2. The four-wire implementation is designed only for category 3 UTP (100Base-T4).
Fig. 2.55: Common Fast Ethernet Implementation Techniques
• Fast Ethernet implementation at the physical layer can be categorized as either two-
wire or four-wire.
2.62
• Manchester encoding scheme needs a 200-Mbaud bandwidth for a data rate of 100
Mbps.
• Three different encoding schemes are used for three different implementations, as
shown in Fig. 2.56.
1. 100Base-TX: It uses two pairs of twisted-pair cable (either category 5 UTP or STP).
For this implementation, the MLT-3 scheme was used. It supports a data rate of
125 Mbps.
2. 100Base-FX: It uses two pairs of fiber-optic cables. 4B/5B encoding is used for this.
It supports 100 to 125 Mbps, which can easily be handled by fiber-optic cable.
3. 100Base-T4: It was designed to use category 3 or higher UTP. The implementation
uses four pairs of UTP for transmitting 100 Mbps. 8B/6T encoding scheme was
used. This means that 100 Mbps uses only (6/8) x 100 Mbps, or 75 Mbaud.
100Base-TX 100Base-FX
4 × 25 Mbps 4 × 25 Mbps 4 × 25 Mbps 4 × 25 Mbps
4B/5B encoder 4B/5B decoder 4B/5B encoder 4B/5B decoder
125 Mbps 125 Mbps 125 Mbps 125 Mbps

MLT-3 encoder MLT-3 decoder NRZ-1 encoder NRZ-1 decoder
Station Station
Two UTP category 5 Two fibers
100Base-T4
100 Mbps 100 Mbps
8B/6T encoder 8B/6T decoder
Station
4 category 3 UTP
Fig. 2.56: Three different Encoding Schemes
2.63
Summary of the Fast Ethernet Implementations:
Sr.
Characteristics 100Base-TX 100Base-FX 100Base-T4
No.
1. Media Cat 5 UTP or STP Fiber Cat 4 UTP
2. Number of wires 2 2 4
3. Maximum length 100 m 185 m 100 m
4. Block encoding 4B/5B 4B/5B Two
5. Line encoding MLT-3 NRZ-I 8B/6T
Advantages of Fast Ethernet:

1. The performance of Fast Ethernet is 10 times more than in traditional Ethernet.
2. Fast Ethernet is easy to set up.
3. Faster throughput for video, multimedia, graphics, Internet surfing, and other
speed-intensive applications.
4. Fast Ethernet supports stronger error detection and correction.
5. Fast Ethernet is ten times faster (100Mbps) than regular 10BaseT networks
(10Mbps).
2.6.6 Gigabit Ethernet

• The IEEE initial standard for Gigabit Ethernet was produced by the IEEE in June 1998
as IEEE 802.3z.
• Gigabit Ethernet operates at 1000 Mbps and supports full-duplex (uses central switch)
and half-duplex approaches (uses CSMA/CD) for medium access.
• Gigabit Ethernet was developed to meet the need for faster communication networks
with applications such as multimedia and Voice over IP (VoIP).
Goals of Gigabit Ethernet:
1. Upgrade the data rate to 1 Gbps.
2. Make it compatible with Standard of Fast Ethernet.
3. Use the same 48-bit address.
4. Keep the same minimum and maximum frame length.
5. Keep the same frame format.
6. To support auto negotiation same as Fast Ethernet.
• Gigabit Ethernet implementations are shown in Fig. 2.57. Two different
implementations of Gigabit Ethernet are two wires and a four Wire.
1. The two-wire implementations use fiber-optic cable (1000Base-SX, short-wave, or
1000Base-LX, long-wave), or STP (1000Base-CX).
2.64
2. The four-wire version uses category 5 twisted-pair cable (1000Base-T).
Fig. 2.57: Gigabit Ethernet Implementations
• Fig. 2.58 shows the encoding/decoding schemes for the four wire implementations of
Gigabit Ethernet.
1000Base-SX, 1000Base-LX, and 1000Base-CX 1000Base-T
8 × 125 Mbps 8 × 125 Mbps 8 × 125 Mbps 8 × 125 Mbps
8B/10B block encoder 8B/10B block decoder

4D-PAM5 encoder 4D-PAM5 decoder
1.25 Gbps 1.25 Gbps
NRZ line encoder NRZ line decoder
Station Station
Two fibers or two STPs 4 UTP cables
Fig. 2.58: The Encoding/Decoding schemes for the Four Wire Implementations
• Gigabit Ethernet uses 8B/10B block encoding, resulting in 1.25 Gbps.

• In the four-wire implementation, 4D-PAM5 encoding is used to reduce the bandwidth.
Thus, all four wires are involved in both input and output; each wire carries 250 Mbps,
which is in the range for category 5 UTP cable.
Summary of Gigabit Ethernet Implementation:
Sr.
Characteristics 1000Base-SX 1000Base-LX 1000Base-CX 1000Base-T
No.
1. Media Fiber short Fiber long STP Cat 5 UTP
wave wave
2. Number of 2 2 2 4
wires
Contd...
2.65
3. Maximum 550 m 5000 m 25 m 100 m

length
4. Block encoding 8B/10B 8B/10B 8B/10B
5. Line encoding NRZ NRZ NRZ 4D-PAM5
Advantages of Gigabit Ethernet:

1. It is roughly 100 times faster than the regular Mbps Ethernet.
2. The elimination of bottlenecks within the Internet service.
3. Improving the traffic flow in overcrowded areas.
4. Transferring large amounts of data across a network quickly.
5. Gigabit Ethernet cables are easier to install and setup.
6. Gigabit Ethernet offers performance enhancement for existing networks without
having to change the cables, protocols and applications already in use.
7. Gigabit Ethernet cable may be a cheaper option than optical fiber.
Disadvantages of Gigabit Ethernet:

1. It is rather expensive.
2. The amount of bandwidth we have is not guaranteed.
Differences between Fast Ethernet and Gigabit Ethernet:
Sr.
Key Fast Ethernet Gigabit Ethernet
No.
1. Successor Fast Ethernet is the Gigabit Ethernet is the
successor of 10-Base-T- successor of Fast Ethernet.
Ethernet.
2. Network speed Fast Ethernet speed is upto Gigabit Ethernet speed is upto
100 Mbps. 1 Gbps.
3. Complexity Fast Ethernet is simple to Gigabit Ethernet is quiet
configure. complex to configure.
4. Delay Fast Ethernet generates Gigabit Ethernet generates
more delay. less delay than Fast Ethernet.
5. Coverage Limit Fast Ethernet coverage limit Gigabit Ethernet coverage
is upto 10KM. limit is upto 70KM.
6. Round trip Fast Ethernet round trip Gigabit Ethernet round trip
delay delay is 100 to 500 bit times. delay is 4000 bit times.
2.66
2.7 WIRELESS LANs (Oct. 17)
• Today, wireless communication is one of the fastest growing technologies. The

demand for connecting devices without the use of cables is increasing everywhere.
• Wireless LANs are those Local Area Networks that use high frequency radio waves
instead of cables for connecting the devices in LAN.
• A wireless LAN (WLAN) is a wireless computer network that links two or more devices
using wireless communication to form a local area network (LAN) within a limited
area such as a home, school, computer laboratory, campus, or office building.
• The two types of Wireless LAN are IEEE 802.11 (sometimes called Wireless Ethernet or
Wi-Fi), and Bluetooth (sometimes called Personal Area Network (PAN)).
• A Wireless Local Area Network (WLAN) links two or more devices over a short
distance using a wireless distribution method, usually providing a connection through
an Access Point (AP) for Internet access.
Fig. 2.59: Wireless LAN
Advantages Wireless LAN:

1. Convenience: All notebook computers and many mobile phones are equipped
with the Wi-Fi technology to connect directly to a wireless LAN.
2. Ease of Setup: Since, a wireless LAN does not require running physical cables
through a location, installation can be quick and cost-effective.
3. Scalability: A wireless LAN can typically expand with existing equipment, while a
wired network might require additional cables and other materials.
4. Security: Controlling and managing access to the wireless LAN is important to its
success. WLANs provide robust security protection, so the data is easily available
to only those users to allow authorised access.
2.67
5. Cost: It can cost less to operate a wireless LAN, which eliminates or reduces wiring
costs during office moves, reconfigurations, or expansions.
6. Mobility: Mobility is a significant advantage of WLANs. Users can access shared
resources without looking for a place to plug in, anywhere in the organization. A
wireless network allows users to be truly mobile as long as the mobile terminal is
under the network coverage area.
Disadvantages of Wireless LAN:

1. Speed: The speed on most wireless LAN networks (typically 1-108 Mbit/s) is
reasonably slow.
2. Security: Wireless LAN are much more susceptible to unauthorized use.
3. Interference: Because wireless LANs use radio signals and similar techniques for
transmission, they are susceptible to interference from lights and electronic
devices.
4. QoS : WLAN offers typically lower QoS. Lower bandwidth due to limitations in
radio transmission and higher error rates due to interference.
2.7.1 Architectural Comparison

• In this section, we compare the architecture of wired and wireless LANs to give some
idea of what we need to look for when we study wireless LANs.
1. Medium:
• In a wired LAN, we use wires/cables to connect hosts. We moved from multiple access
to point-to-point access through the generation of the Ethernet.
• In a wireless LAN, the medium is air, the signal is generally broadcast. When hosts in a
wireless LAN communicate with each other, they are sharing the same medium
(multiple access).
2. Host:
• In a wired LAN, a host is always connected to its network at a point with a fixed link
layer address related to its Network Interface Card (NIC).
• A host can move from one point in the Internet to another point. In this case, its link-
layer address remains the same, but its network-layer address will change.
• However, before the host can use the services of the Inter- net, it needs to be physically
connected to the Internet.
• In a wireless LAN, a host is not physically connected to the network; it can move freely
and can use the services provided by the network.
• Therefore, mobility in a wired network and wireless network are totally different
issues.
2.68
3. Connected to other Networks:

• A wired LAN can be connected to another network or an internetwork such as the
Internet using a router.
• A wireless LAN may be connected to a wired infrastructure network, to a wireless
infrastructure network, or to another wireless LAN.
• Fig. 2.60 shows the two environments of a wired LAN and a wireless LAN.
Fig. 2.60: Connection of a Wired and a Wireless LAN to other Networks
• In this case, the wireless LAN is referred to as an infrastructure network and the
connection to the wired infrastructure, such as the Internet, is done via a device called
an Access Point (AP).
• An AP is gluing two different environments together namely, one wired and one
wireless.
• Communication between the AP and the wireless host occurs in a wireless
environment while communication between the AP and the infrastructure occurs in a
wired environment.
4. Isolated LANs:
• The concept of a wired isolated LAN also differs from that of a wireless isolated LAN.
A wired isolated LAN is a set of hosts connected via a link-layer switch (in the recent
generation of Ethernet).
• A wireless isolated LAN, called an ad for network in wireless LAN terminology, is a set
of hosts that communicate freely with each other.
Fig. 2.61: Isolated LANs (Wired vs Wireless)
2.69
• The concept of a link-layer switch does not exist in wireless LANs. Fig. 2.61 shows two
isolated LANs, one wired and one wireless.
5. Moving between Environments:
• A wired LAN or a wireless LAN operates only in the lower two layers of the TCP/IP
protocol suite.
• Means that if we have a wired LAN in a building that is connected via a router or a
modem to the Internet, all we need in order to move from the wired environment to a
wireless environment is to change the network interface cards designed for weed
environments to the ones designed for wireless environments and replace the link-
layer switch with an access point.
• In this change, the link-layer addresses will change (because of changing NICs), but the
network-layer addresses (IP addresses) will remain the same; we are moving from
wired links to wireless links.
2.7.2 Characteristics
• There are a number of characteristics of wireless LANs that either do not apply to
wired LANs or the existence of which is negligible and can be ignored.
• Some of these characteristics are explained below:
1. Interference:
• The main issue is that a receiver may receive signals not only from the intended
sender, but also from other senders if they are using the same frequency band.
2. Attenuation:
• The strength of electromagnetic signals decreases rapidly because the signal disperses
in all directions; only a small portion of it reaches the receiver.
• The situation becomes worse with mobile senders that operate on batteries and
normally have small power supplies.
3. Error:
• The errors and error detection are more serious issues in a wireless network than in a
wired network.
4. Multipath Propagation:
• A receiver may receive more than one signal from the same sender because
electromagnetic waves can be reflected back from obstacles such as walls, the ground,
or objects.
• The result is that the receiver receives some signals at different phases (because they
travel different paths). This makes the signal less recognizable.
2.70
2.7.3 Access Control

• In wireless LAN, CSMA/CD cannot be implemented because of following reasons:
1. For collision detection, a station must be able to send data and receive collision
signals at the same time, which increases bandwidth requirements.
2. Collisions may not be detected because of the hidden station problem.
3. The distance between stations can be great. Signal fading could prevent a station at
one end from hearing a collision at the other end.
• To overcome the above three problems, Carrier Sense Multiple Access with Collision
Avoidance (CSMA/CA) was invented for wireless LANs.
• Fig. 2.62 shows data exchange and control frames used in CSMA/CA. Before sending a
frame, the source station senses the medium. There are two options:
1. The channel uses a persistence strategy with back-off until the channel is idle.
2. After the station is found to be idle, the station waits for a period of time called the
Distributed InterFrame Space (DIFS), then the station sends a control frame
Request To Send (RTS).
• After receiving the RTS and waiting for a period Short InterFrame Space (SIFS), the
destination sends a control frame called Clear To Send (CTS). This control frame
indicates to source that the destination station is ready to receive data.
• The source then waits for time equal to SIFS and then sends data. After receiving data,
the destination waits for time equal to SIFS and sends acknowledgement to the source.
Network Allocation Vector (NAV):

• It’s interesting to see how the collision avoidance is handled by the protocol. CSMA/CA
protocol uses a feature called NAV.
• When a station sends to an RTS frame, it also includes the total time that is needed to
occupy the channel.
• The stations that are affected by this transmission create a timer called NAV, that
shows how much time these stations should not sense the channel.
• Each time a station accesses the system and sends to an RTS frame, another station
starts NAV.
• Two or more stations may try to send RTS frames at the same time. These control
frames may collide and destroy.
2.71
• The sender assumes there has been a collision if it has not received a CTS from the
receiver. In such a situation, the sender tries again. This concept is known as collision
during handshaking.
Start
Set back-off
to zero
Persistence
strategy
Wait DIFS
Send RTS
Wait back-off
time Set a timer
No CTS received
before time-out ?
Yes
Wait SIFS
Send the frame
Set a timer
No
Back-off Increment No ACK received

limit ? back-off before time-out ?
Yes Yes
Abort Success
Fig. 2.62: Data Exchange and Control Frames used in CSMA/CA
2.72
2.7.4 IEEE 802.11 Architecture (April 16)
• IEEE defines a standard for wireless LAN, named IEEE 802.11, which covers the
physical and data link layers.
• IEEE has defined the specifications for a wireless LAN, called IEEE 802.11, which
covers the physical and data-link layers.
• It is sometimes called wireless Ethernet. In some countries, including the United
States, the public uses the term WiFi (Wireless Fidelity) as a synonym for wireless
LAN.
• WiFi however, is a wireless LAN that is certified by the WiFi Alliance, a global,
nonprofit industry association of more than 300 member companies devoted to
promoting the growth of wireless LANs.
• The IEEE 802.11 standard defines two different services BSS and ESS as explained
below:
1. Basic Service Set (BSS): (April 18; Oct. 18)
• BSS is a building block for a Wireless LAN. A BSS is made up of stationary or mobile
wireless station and an optional central base station, known as Access Point (AP).
• The BSS without an AP is a stand-alone network and called as ad hoc architecture.
Such types of networks cannot send data to other BSSs.
• Stations can form a network without the need of AP. Stations can locate one another
and agree to be part of a BSS.
• A BSS with an AP is called an infrastructure network. All stations in such architecture
are communicating through an AP.
• Fig. 2.63 shows two sets of IEEE 802.11 standard.
BSS : Basic Service Set
AP : Access Point
Station Station Station Station
AP
Station Station Station Station
Ad hoc network (BSS without an AP) Infrastructure (BSS with an AP)
Fig. 2.63: Two Sets of IEEE 802.11 Standard

2.73
2. Extended Service Set (ESS):

• An extended service set is made up of two or more BSSs with APs.
• The BSSs are connected through a distribution system, which is a wired LAN. The
distribution system connects the APs in the BSSs.
• ESS uses two types of stations, i.e., mobile and stationary stations. The mobile stations
are the normal stations in the BSS. The stationary stations are AP stations that are part
of a wired LAN.
• Communication between two stations from two different BSSs take place via., two APs.
But when stations are within reach of one another, they can communicate directly,
without the use of AP.
Distribution system
Server or
Gateway
ESS
AP AP AP
BSS BSS BSS
Fig. 2.64: Extended Service Set (ESS)
Station Types:
• Based on mobility, IEEE 802.11 defines three types of stations in a wireless LAN, as
given below:
1. No-transition: A station with no-transition mobility is either stationary (not
moving) or moving only inside a BSS.
2. BSS-transition: A station with BSS-transition mobility can move from one BSS to
another, but the movement is in one ESS only.
3. ESS-transition: A station with ESS-transition mobility can move from one ESS to
another.
2.74
2.7.5 Physical Layer (April 18)
• As we know that the physical layer is responsible for converting data streams into
signals, the bits of 802.11 networks can be converted to radio waves or infrared waves.
• The 802.11 physical layer (PHY) is the interface between the MAC and the wireless
media where frames are transmitted and received.
• The specifications of IEEE 802.11 are explained below:
1. IEEE 802.11 Infrared:
• It uses diffused infrared light in the range of 800 to 950 nm. It allows two different
speeds 1 Mbps and 2Mbps.
• For a 1 Mbps data rate, 4 bits of data are encoded into 16 bit code. This 16 bit code
contains fifteen as and a single 1.
• For a 2-Mbps data rate, a 2 bit code is encoded into 4 bit code. This 4 bit code contains
three 0s and a single 1.
• The modulation technique used is Pulse Position Modulation (PPM) i.e. for converting
digital signal to analog.
2. IEEE 802.11 FHSS:
• IEEE 802.11 uses Frequency Hopping Spread Spectrum (FHSS) method for signal
generation.
• The FHSS method uses a 2.4 GHz ISM band. This band is divided into 79 sub-bands of
1MHz with some guard bands.
• In this method, at one moment data is sent by using one carrier frequency and then by
some other carrier frequency at the next moment.
• After this, idle time is there in communication. This cycle is repeated after regular
intervals.
• In FHSS a pseudo random number generator selects the hopping sequence. The
allowed data rates are 1 or 2 Mbps.
• The FHSS method uses frequency shift keying (two level or four level) for
modulation i.e. for converting digital signal to analogy.
3. IEEE 802.11 DSSS:
• This method uses Direct Sequence Spread Spectrum (DSSS) method for signal
generation. Each bit is transmitted as 11 chips using a Barker sequence.
• DSSS uses the 2.4-GHz ISM band. It also allows the data rates of 1 or 2 Mbps.
• It uses Phase Shift Keying (PSK) technique at 1 M baud for converting digital signal to
analog signal.
2.75
4. IEEE 802.11a OFDM:

• This method uses Orthogonal Frequency Division Multiplexing (OFDM) for signal
generation.
• This method is capable of delivering data upto 18 or 54 Mbps. In OFDM all the
subbands are used by one source at a given time.
• It uses a 5 GHz ISM band. This band is divided into 52 subbands, with 48 subbands for
data and 4 subbands for control information.
• If Phase Shift Keying (PSK) is used for modulation then data rate is 18 Mbps. If
Quadrature Amplitude Modulation (QAM) is used, the data rate can be 54 Mbps.
5. IEEE 802.11b HR-DSSS:
• It uses the High Rate Direct Sequence Spread Spectrum method for signal generation.
• HR-DSSS is similar to DSSS except for the encoding method. Here, 4 or 8 bits are
encoded into a special symbol called Complementary Code Key (CCK).
• It uses a 2.4 GHz ISM band. It supports four data rates 1, 2, 5.5 and 11 Mbps. 1 Mbps
and 2 Mbps data rates use phase shift modulation. The 5.5 Mbps version uses BPSK
and transmits at 1.375 Mbaud/s with 4-bit CCK encoding.
• The 11 Mbps version uses QPSK and transmits at 1.375 Mbps with 8-bit CCK encoding.
6. IEEE 802.11g OFDM:
• It uses OFDM modulation technique. It uses a 2.4 GHz ISM band.
• It supports the data rates of 22 or 54 Mbps. It is backward compatible with 802.11 b.
7. IEEE 802.11n:
• An upgrade to the 802.11 project is called 802.11n (the next generation of wireless
LAN). The goal is to increase the throughput of 802.11 wireless LANs.
• The new standard emphasizes not only the higher bit rate but also eliminating some
unnecessary overhead. The standard uses what is called MIMO (Multiple-Input
Multiple-Output antenna) to overcome the noise problem in wireless LANs.
2.7.6 MAC Sublayer

• IEEE 802.11 defines two MAC sublayers, Distribution Coordination Function (DCF) and
Point Coordination Function (PCF).
• Fig. 2.65 shows the relationship between the two MAC sublayers, the LLC sublayer and
the physical layer.
• CSMA/CA (Carrier Sense Multiple Access with Collision Avoidance) is the protocol used
to access methods defined by IEEE at the MAC sublayer is called the Distribution
Coordination Function (DCF).
2.76
• The Point Co-ordination Function (PCF) is an optional access method which is

implemented in an infrastructure network and not in an ad hoc network. It is
implemented on top of the DCF and is used for time sensitive transmission.
LLC
sublayer IEEE 802.1
Contention-free service
Contention
Data link service
layer
Point Co-ordination Function (PCF)
MAC
sublayer
Distributed Co-ordination Function (DFC)
802.11 802.11 802.11 802.11a 802.11a 802.11g

Physical layer FHSS DSSS Infrared DSSS OFDM DSSS
Fig. 2.65: The Relationship between Layers
2.7.6.1 Frame Format (Oct. 17)
• The MAC layer consists of nine fields as shown in Fig. 2.66.

2 bytes 2 bytes 6 bytes 6 bytes 6 bytes 2 bytes 6 bytes 0 to 2312 bytes 4 bytes
Address Address Address Address Frame

FC D 1 2 3 SC 4 body FCS
Protocol To From More Power More

Type Subtype Retry WEP Rsvd
version DS DS flag mgt. data
2 bits 2 bits 4 bits 1 bit 1 bit 1 bit 1 bit 1 bit 1 bit 1 bit 1 bit
Fig. 2.66: MAC Layer Frame Format
• The MAC layer frame consists of following nine fields:

1. Frame Control (FC): The FC field is 2 bytes long and defines the type of frame and
some control information. The subfields of FC field are listed below:
(i) Version: Current version is 0.
2.77
(ii) Type: Type of information: management (00), control (01), or data (10).
(iii) Subtype: Subtype of each type.
(iv) To DS: Defined later.
(v) From DS: Defined later.
(vi) More flag: When set to 1, means more fragments.
(vii) Retry: When set to 1, means retransmitted frame.
(viii) Pwr mgt: When set to 1, means the station is in power management mode.
(ix) More data: When set to 1, means the station has more data to send.
(x) WEP: Wired equivalent privacy (encryption implemented).
(xi) Rsvd: Reserved.
2. D: In all frames, this field defines the duration of the transmission that is used to
set the value of NAV.
3. Addresses: There are four address fields, each 6 bytes long. The meaning of each
address field depends on the value of the To DS and From DS subfields.
To DS From DS Address 1 Address 2 Address 3 Address 4
0 0 Destination Source BSS ID N/A
0 1 Destination Sending AP Source N/A
1 0 Receiving AP Source Destination N/A
1 1 Receiving AP Sending AP Destination Source

4. Sequence control: This field defines the sequence number of the frame to be used
in flow control.
5. Frame body: This field between 0 to 2312 bytes contains information based on the
type and the subtype defined in the FC field.
6. FCS: This field is used for error detection.
Frame Types:
• IEEE 802.11, wireless LAN defines three types of frames, as explained below:
1. Management Frames: Management frames are used for the initial
communication between stations and access points.
2. Control Frames: Control frames are used for accessing the channel and
acknowledging frames.
3. Data Frames: Data frames are used for carrying data and control information.
2.78
2.7.7 Bluetooth (April 17)
• Bluetooth is a wireless LAN technology used to connect devices of different functions

such as telephones, computers like laptop or desktop, notebooks, cameras, printers
and so on.
• The Bluetooth specifications are developed and licensed by the Bluetooth Special
Interest Group.
• Bluetooth technology is a short-range wireless radio technology that allows electronic
devices to connect to one another.
• Generally, Bluetooth has a range of up to 30 ft., or greater, depending on the Bluetooth
Core Specification Version. Newer devices, using newer versions of Bluetooth, have
ranges over 100 ft.
• A Bluetooth LAN is an ad hoc network, which means that the network is formed
spontaneously. The devices find each other and make a network.
• Bluetooth is an open specification for short-range wireless transmission of voice and
data. It provides a simple, low-cost seamless wireless connectivity between portable
handheld devices.
• Bluetooth technology is the implementation of a protocol defined by the IEEE 802.15
standard.
• The standard defines a wireless Personal Area Network (PAN) operable in an area the
size of a room or a hall.
2.7.7.1 Bluetooth Architecture (Oct. 18)
• Bluetooth architecture defines two types of networks i.e., Piconet and Scatternet.
1. Piconet: (Oct. 18)

• Piconet is a Bluetooth network that consists of one primary (master) node and seven
active secondary (slave) nodes.
• Thus, piconet can have upto eight active nodes (1 master and 7 slaves) or stations
within the distance of 10 meters.
• There can be only one primary or master station in each piconet.
• The communication between the primary and the secondary can be one-to-one or one-
to-many.
• Fig. 2.67 shows typical piconet. All communication is between master and a slave.
Salve-slave communication is not possible.
• In addition to seven active slave stations, a piconet can have up to 255 parked nodes.
These parked nodes are secondary or slave stations and cannot take part in
communication until it is moved from parked state to active state.
2.79
Primary/master
Secondary/Slave Secondary/Slave Secondary/Slave Secondary/Slave
Fig. 2.67: Piconet
2. Scatternet: (April 18)

• Scattemet is formed by combining various piconets.
• A slave in one piconet can act as a master or primary in another piconet.
• Fig. 2.68 shows scatternet.
Scatternet
Primary
Secondary
Primary
Secondary Secondary
Secondary Secondary
Piconet 1 Bridge slave Piconet 2
Fig. 2.68: Scatternet
• Such a station or node can receive messages from the master in the first piconet and
deliver the message to its slaves in other piconet where it is acting as master. This
node is also called bridge slave.
2.80
• Thus, a station can be a member of two piconets. A station cannot be a master in two
piconets.
2.7.7.2 Bluetooth Layers

• Bluetooth network technology connects mobile devices wirelessly over a short-range
to form a Personal Area Network (PAN).
• The Bluetooth architecture, showing all the major layers in the Bluetooth system, are
shown in Fig. 2.69.
• The layers below can be considered to be different hurdles in an obstacle course. This
is because all the layer’s function one after the other. One layer comes into play only
after the data has been through the previous layer.
Fig. 2.69: Bluetooth Architecture
• Layers of Bluetooth in Fig. 2.69 are explained below:

1. Radio Frequency (RF): The radio frequency layer defines the requirements for a
Bluetooth transceiver operating in the 2.4 GHz ISM band.
2. Baseband: The baseband layer describes the specification of the Bluetooth Link
Controller (LC), which carries out the base band protocols and other low-level link
routines. It specifies piconet/channel definition, "low-level" packet definition,
channel sharing.
3. LMP: The Link Manager Protocol (LMP) is used by the link managers (on either
side) for link set-up and control.
4. HCI: The Host Controller Interface (HCI) provides a command interface to the
baseband link controller and link manager, and access to hardware status and
control registers.
2.81
5. L2CAP: Logical Link Control and Adaptation Protocol (L2CAP) supports higher
level protocol multiplexing, packet segmentation and reassembly, and the
conveying of quality of service information.
6. Radio Frequency Communication (RFCOMM): RFCOMM layer takes care of the
communication channel between two devices or between a master and a slave. It
connects the serial ports of all the devices according to the requirement.
7. Service Discovery Protocol (SDP): This layer provides a means for applications to
discover which services are available and to determine the characteristics of those
available services.
8. Telephony Control protocol Specification (TCS): Basic function of this layer is
call control (setup and release) and group management for gateway serving
multiple devices.
9. Application Program Interface (API) Libraries: These are software modules
which connect the host application program to the Bluetooth communications
system. As such they reside and execute on the same processing resource as the
host system application.
2.7.7.3 Bluetooth Frame Format

• Fig. 2.70 shows the frame format of Bluetooth.
Fig. 2.70
• The various fields of Bluetooth frame format are explained below:
1. Access Code: It is a 72 bit field that contains synchronization bits. It identifies the
master.
2. Header: This is a 54-bit field. It contains an 18 bit pattern that is repeated for 3
times. The header field contains following subfields:
(i) Address: This 3 bit field can define up to seven slaves (1 to 7). If the address
is zero, it is used for broadcast communication from primary to all
secondaries.
2.82
(ii) (ii)Type: This 4 bit field identifies the type of data coming from upper layers.
(iii) F: This flow bit is used for flow control. When set to 1, it means the device is
unable to receive more frames.
(iv) A: This bit is used for acknowledgement.
(v) S: This bit contains a sequence number of the frame to detect retransmission.
As stop and wait protocol is used, one bit is sufficient.
(vi) Checksum: This 8 bit field contains checksum to detect errors in header.
3. Data: This field can be 0 to 2744 bits long. It contains data or control information
coming from upper layers.
2.7.7.4 Advantages, disadvantages and Applications of Bluetooth

Advantages of Bluetooth:
1. It is cheaper in cost.
2. Easy to install and setup.
3. It makes connecting to different devices convenient.
4. Setting up a Bluetooth connection between two devices is quick and easy.
5. It is wireless technology.
6. Bluetooth doesn't need any configuration to start a connection and perform file
transfers.
7. Bluetooth is actually inexpensive.
8. Bluetooth is automatic i.e., when two or more devices enter a range of up to 30 feet
of each other, they will automatically begin to communicate without you having to
do anything.
9. Bluetooth devices almost always avoid interference from other wireless devices.
10. The standard for Bluetooth will allow compatible devices to share data and voice
communications. This is great for mobile phones and headsets.
Disadvantages of Bluetooth:
1. Bluetooth only allows short range communication between devices (less than 10
meters).
2. Bluetooth only offers 1 Mbps to 3 Mbps data transfer rate.
3. Bluetooth makes it much more open to interception and attack. For this reason,
security is a very key aspect to the Bluetooth specification.
4. It can lose connection in certain conditions.
Applications of Bluetooth: (April 17; Oct. 17)
1. It is used for providing communication between peripheral devices like wireless
mouse or keyboard with the computer.
2.83
2. It is used by modern healthcare devices to send signals to monitors.

3. It is used by modern communicating devices like mobile phones, PDAs, palmtops
etc. to transfer data rapidly.
4. It is used for dial up networking. Thus allowing a notebook computer to call via a
mobile phone.
5. It is used for cordless telephoning to connect a handset and its local base station.
6. It also allows hands-free voice communication with a headset.
7. It also enables a mobile computer to connect to a fixed LAN.
Difference between Wireless LAN and Bluetooth: (April 19)
Sr.
Bluetooth Wireless LAN
No.
1. It is a short range technology It refers to a network that connects two
standard which allows devices to or more devices by using wireless data
communicate in a wireless manner. connections over short distances.
2. The distance range is 30 feet to 100 The distance range is Up to 400 feet.
feet.
3. It requires low bandwidth (not for It requires high bandwidth.
transferring large files).
4. Bluetooth has generally lower speed. Wireless LAN is much faster compared to
Bluetooth.
5. Lower cost. Cost is much more expensive than
Bluetooth.
6. Bluetooth chips have lower power It requires more power consumption.
consumption - less drain on battery.
7. It is less secure. More secure.
8. Fairly simple to use. Can be used to It is more complex and requires
connect upto seven devices at a time. configuration of hardware and software.
It is easy to switch between devices or
find and connect to any device.
PRACTICE QUESTIONS
Q. I Multiple Choice Questions:
1. Which are the following lower layers of the OSI model?
(a) Physical layer (b) Data link layer
2.84
2. Which layer is concerned with transmission of raw bits over a communication

channel/medium?
(c) Network layer (d) All of these
3. Which is responsible for the reliable transfer of data frames from one node to
another connected by the physical layer?
(c) Network layer (d) All of these
4. The two sublayers of the data link layer are ______.
(a) Logical Link Control (LLC) (b) Medium Access Control (MAC)
5. The design issues of the data link layer are ______.
(a) Controlling over the flow of data
(b) Providing the service interface to the network layer
(c) Dealing with the transmission errors
(d) All of these
6. Breaking the bit stream into frames is called ______.
(a) Packetizing (b) Framing
7. Which is a state occurring in the network layer when the message traffic is so
heavy that it slows down network response time?
(a) Congestion (b) Error
(c) Flow (d) All of these
8. A link layer address is sometimes called as ______.
(a) a link address (b) a physical address
(c) a MAC address (d) All of these
9. Which method uses a field in the header to specify the number of characters in the
frame?
(a) Character count (b) Flag bytes with byte stuffing
(c) Starting and ending flags (d) None of these
10. Which layer provides addressing and channel access control mechanisms that
make it possible for several stations to communicate within a multiple access
network that incorporates a shared medium?
(a) LLC (b) MAC
2.85
11. Which is a process in which a single channel is divided and allotted to multiple
users in order to carry user specific tasks.
(a) Data allocation (b) Communication allocation
(c) Channel allocation (d) All of these
12. In which channel allocation, frequency bands are not permanently assigned to the
users?
(a) Dynamic (b) Static
(c) Fixed (d) All of these
13. Which is a system for coordinating and arbitrating access to a shared
communication Networks channel?
(a) CSMA (b) FDMA
(c) ALOHA (d) None of these
14. Which is a network access method used on shared network topologies such as
Ethernet to control access to the network?
(a) CSMA (b) FDMA
(c) ALOHA (d) None of these
15. In which method, the station needs to make a "before sending data".
(a) Polling (b) Reservation
(c) Select (d) None of these
16. Which is the multiple access method in which the available bandwidth of a link is
shared in time, frequency or through code between different stations?
(a) Polling (b) Reservation
(c) Channelization (d) None of these
17. Which is a channel access method for stored medium networks?
(a) TDMA (b) FDMA
(c) CDMA (d) All of these
18. A switched network consists of a series of interlinked nodes called ______.
(a) Switches (b) Routers
(c) Repeaters (d) All of these
19. Which switching takes place at the physical layer of the TCP/IP reference model?
(a) Message (b) Packet
(c) Circuit (d) None of these
20. Virtual circuit packet switching is normally done at the ______ layer.
(a) Physical (b) Data link
(c) Network (d) None of these
2.86
21. Which is a standardized system for connecting computers to a LAN?

(a) Twisted pair (b) Coaxial cable
(c) Ethernet (d) None of these
22. Which standard defines rules for configuring an Ethernet network?
(a) 802.3 (b) 802.3u
(c) 802.ba (d) None of these
23. Which is a type of contention oriented protocol that defines how to respond when
a collision is detected?
(a) CSMA (b) CSMA/CA
(c) CSMA/CD (d) None of these
24. Which standard is defined for wireless LAN?
(a) IEEE 802.3 (b) IEEE 802.11
(c) IEEE 802.ba (d) IEEE 802.3u
25. A ______ with an AP is called an infrastructure network.
(a) BSS (b) DS
(c) ESS (d) None of these
26. Which is a short-range (below 30 ft.) wireless radio technology that allows
electronic devices to connect to one another?
(a) WiMax (b) Bluetooth
(c) WiFi (d) None of these
ANSWERS
1. (c) 2. (a) 3. (b) 4. (c) 5. (d) 6. (b) 7. (a)
8. (d) 9. (a) 10.(b) 11. (c) 12. (a) 13. (c) 14. (a)
15. (b) 16. (c) 17. (a) 18. (a) 19. (c) 20. (b) 21. (c)
22. (a) 23. (c) 24. (b) 25. (a) 26. (b)

1. The ______ layer provides services to data link layer.
2. The ______ provides for the transfer of data frames between hosts connected to the
physical link.
3. Signals can be either ______ or ______.
4. Digital data refers to information that has ______ states.
5. The maximum rate at which data can be correctly communicated over a channel
in presence of noise and distortion is known as its ______ capacity.
2.87
6. The term ______ refers to the speed of data transmissions.

7. ______ is the rate of successful message delivery over a communication
channel/medium.
8. ______ is the time it takes for a packet to get across the network, from source to
destination.
9. The ______ product defines the number of bits that can fill the network link.
10. ______ is the variance in time delay in milliseconds (ms) between data
packets over a network. It is a disruption in the normal sequence of sending data
packets.
11. The ______ layer is located between the physical and the network layers.
12. ______ control design issue deals with transmission errors.
13. In ______ connectionless service, the sender machine sends the frames to the
destination machine without having the destination machine acknowledge them.
14. Group of physical layer bits, stream into units (messages) called as ______.
15. ______ in the data link layer separates a message from one source to a destination
by adding a sender address and a destination address.
16. ______ allocation may be done using two ways Static Channel Allocation and
Dynamic Channel Allocation.
17. The traditional way of allocation the single channel for______ users is the
Frequency Division Multiplexing (FDM).
18. Media access methods are implemented at the ______ layer.
19. ______ was designed for a wireless LAN, but it can be used on any shared medium.
20. A station is allowed to send data when the station receives a special frame called
______.
21. In ______, the stations are organized in a logical ring.
22. In ______, the entire bandwidth is one channel.
23. At the ______ layer of the TCP/IP protocol suite, we can have only circuit switching.
24. At the data link layer of the TCP/IP protocol suite, we can have ______ switching.
25. In packet switching, each packet has ______ and destination addresses, travelling
from one point (router) to the other point (router).
26. In packet switching, ______ are divided into packets of fixed or variable size.
27. A ______ is simply a collection of two or more computers, printers, and other
devices linked by Ethernet cables.
28. Ethernet provides a ______ service which means each frame sent is independent of
the previous or next frame.
2.88
29. ______ Ethernet is an Ethernet standard for 100-Mbps data transmission defined
by the IEEE 802.3u specification.
30. The IEEE initial standard for ______ Ethernet was produced by the IEEE in June
1998 as IEEE 802.3z.
31. Wireless LANs are those Local Area Networks that use high frequency ______
waves instead of cables for connecting the devices in LAN.
32. A wired ______ LAN is a set of hosts connected via a link-layer switch.
33. An ______ is made up of two or more BSSs with APs.
34. A Bluetooth LAN is an ______ network.
35. ______ is formed by combining various piconets.
36. In the data link layer, ______ control restricts the number of frames the sender can
send before it waits for an acknowledgment from the receiver.
37. ______ count method uses a field in the header to specify the number of characters
in the frame.
38. ______ method allows data frames to contain an arbitrary number of bits and
allows character codes with an arbitrary number of bits per character.
39. ______ channel allocation are schemes for allotting shared network channels to
competing users in a dynamic manner as per their requirements.
40. The Bluetooth ______ layer performs functions similar to those in the Internet
model’s MAC sublayer.
ANSWERS
1. Physical 2. Data link 3. Analog, digital 4. Discredit
5. Channel 6. Bandwidth 7. Throughput 8. Latency
9. Bandwidth-delay 10. Jitter 11. Data link 12. Error
13. Unacknowledged 14. Frames 15. Framing 16. Channel
17. Multiple 18. Data link 19. Aloha 20. Token
21. Token passing 22. TDMA 23. Physical 24. Packet
25. Source 26. Messages 27. Wired network 28. Connectionless
(LAN)
29. Fast 30. Gigabit 31. Radio 32. Isolated
33. ESS 34. Ad hoc 35. Scattemet 36. Flow
37. Character 38. Bit stuffing 39. Dynamic 40. Baseband
2.89

1. The physical layer (1ayer 1) of OSI and TCP/IP models deals with transmission of
individual bits from one node to another over a physical medium.
2. Digital signals have infinite values in a range and Digital Signals have a limited
number of defined values.
3. Digital data takes on discrete values. For example, data is stored in computer
memory in the form of 0s and 1s.
4. Propagation time measures the time require for a bit to travel from the source to
destination.
5. The data link layer is layer number 2 in the ISO-OSI and TCP/IP model.
6. Data link layer provides the mechanism of error control in which it detects and
retransmits damaged or lost frames.
7. The data link layer divides the stream of bits received from the network layer into
manageable data units called packets.
8. The data link layer ensures error free link for data transmission.
9. A link is controlled at the data link layer, the addresses need to belong to the data
link layer.
10. In static channel allocation, a fixed portion of the frequency channel is allotted to
each competing user.
11. CSMA/CD stands for Carrier Sense Multiple Access with Collision Detection.
12. CSMA/CA stands for Carrier Sense Multipoint Access with Collision Avoidance.
13. In FDMA (Frequency Division Multiple Access), the bandwidth is shared by all
stations.
14. In CDMA (Code Division Multiple Access), only one channel occupies the entire
bandwidth of the link. All stations can send data simultaneously.
15. FDMA is an access method in the physical layer.
16. In packet switching, each packet has source and destination addresses, travelling
from one point (router) to the other point (router).
17. In Virtual Circuit Packet Switching, packets are sents in sequential order over a
defined route.
18. In message switching, it is not necessary to establish a dedicated path between
transmitter and receiver.
19. The IEEE has subdivided the data link layer into two sub layers i.e. LLC (Logical
Link Control) and MAC (Media Access Control).
2.90
20. A wireless isolated LAN, called an ad for network in wireless LAN terminology, is
a set of hosts that communicate freely with each other.
21. A ESS is made up of stationary or mobile wireless station and an optional central
base station, known as Access Point (AP).
22. IEEE 802.11 defines two MAC sublayers, Distribution Co-ordination Function
(DCF) and Point Co-ordination Function (PCF).
23. The Point Co-ordination Function (PCF) is an optional access method which is
implemented in an infrastructure network and not in ad hoc network.
24. Piconet is a Bluetooth network that consists of one primary (master) node and
seven active secondary (slave) nodes.
25. Bandwidth describes the volume of data that can be transferred at a given time.
26. Throughput is the actual amount of data that is successfully sent/received over
the communication link.
27. Jitter variation in packet delay at the receiver of the information.
28. Throughput refers to measurement of data transferred in a specific time period.
29. Flow control in the data link layer is the process of detecting and correcting data
frames that have been corrupted or lost during transmission.
30. In the dynamic channel allocation scheme, a fixed portion of the frequency
channel is allotted to each user.
31. In 1-persistent CSMA method, a station that wants to transmit data continuously
senses the channel to check whether the channel is idle or busy.
32. In controlled access, the stations consult one another to find which station has the
right to send.
33. Packet switching is a method of transferring the data to a network in form of
packets.
34. A Gigabit Ethernet network can transmit data at a rate up to 10 Megabits per
second (10 Mbps).
35. The Fast Ethernet standard (IEEE 802.3u) has been established for Ethernet
networks that need higher transmission speeds. This standard raises the Ethernet
speed limit from 10 Mbps to 100 Mbps.
36. There are three types of Fast Ethernet: 100BASE-TX for use with level 5 UTP cable;
100BASE-FX for use with fiber-optic cable; and 100BASE-T4 which utilizes an extra
two wires for use with level 3 UTP cable. The 100BASE-TX standard has become
the most popular due to its close compatibility with the 10BASE-T Ethernet
standard.
2.91
37. Gigabit Ethernet (GbE) is the family of Ethernet technologies that achieve
theoretical data rates of 1 gigabit per second (1 Gbps).
38. Wired LANs are wireless computer networks that use high-frequency radio waves
instead of cables for connecting the devices within a limited area forming LAN.
39. The IEEE 802.11 standard for wireless LANs defines two services: basic service set
(BSS) and extended service set (ESS).
40. The physical layer methods used by wireless LANs include frequency-hopping
spread spectrum (FHSS), direct sequence spread spectrum (DSSS), orthogonal
frequency-division multiplexing (OFDM), and high-rate direct sequence spread
spectrum (HR-DSSS).
41. The network allocation vector (NAV) is a timer used for collision avoidance.
42. The Bluetooth radio layer performs functions similar to those in the Internet
model’s physical layer.
ANSWERS
1. (T) 2. (F) 3. (T) 4. (T) 5. (T) 6. (T)
7. (F) 8. (T) 9. (T) 10. (T) 11. (F) 12. (F)
13. (F) 14. (T) 15. (F) 16. (T) 17. (T) 18. (T)
19. (T) 20. (T) 21. (F) 22. (T) 23. (T) 24. (F)
25. (T) 26. (T) 27. (T) 28. (T) 29. (F) 30. (F)
31. (T) 32. (T) 33. (T) 34. (F) 45. (T) 36. (T)
37. (T) 38. (F) 39. (T) 40. (T) 41. (T) 42. (T)

1. List lower layers in OSI model.
2. What is bandwidth?
3. Define the term latency (delay).
4. What is throughput?
5. Define the term framing?
6. What is jitter?
7. List design issues of data link layer.
8. What is a frame?
9. List services of data link layer.
10. What is flow control?
2.92
11. Define congestion?

12. List various framing methods.
13. What is meant by channel allocation?
14. What is ALOHA?
15. What is CSMA?
16. What is CSMA/CA?
17. What is CSMA/CD?
18. Define polling.
19. What is meant by channelization?
20. What is TDMA and FDMA?
21. Define switching.
22. List types of switching.
23. Give approaches for packet switching.
24. Define the term wired LANs.
25. What is Ethernet?
26. What is wireless LAN?
27. List components for IEEE 802.11 architecture.
28. What is Bluetooth?
29. List components for Bluetooth architecture.
1. With the help of diagrams describe communication at the physical layer.
2. Write Nyquist and Shannon's formula for calculating data rate of a channel.
3. Define framing? Explain frame format with its different fields.
4. Describe error control in detail.
5. Explain the term congestion control in detail.
6. Write short note on: Link layer addressing.
7. With a suitable diagram describe the character count method.
8. Describe the term flag bytes with byte stuffing.
9. Explain physical layer coding violations in detail.
10. What is static and dynamic channel allocation? Compare them.
11. Explain slotted and pure ALOHA.
12. Compare CSMA/CA and CSMA/CD.
13. Describe access control diagrammatically.
2.93
14. Explain the term token passing in detail.

15. Define channelization. Also explain FDMA and TDMA.
16. Explain packet switching diagrammatically.
17. What is circuit switching? How does it work? State its advantages and
disadvantages.
18. With the help of diagram describe message switching.
19. Differentiate between circuit, packet and message switching.
20. With the help of example describe datagram packet switching.
21. With the help of example, describe virtual circuit packet switching.
22. Explain wired and wireless LANs. Also compare them.
23. Explain following Ethernet with implementations:
(i) Fast Ethernet
(ii) Gigabit Ethernet.
24. Explain isolated LANs diagrammatically.
25. Describe IEEE 802.11 architecture in detail.
26. What are the responsibilities of a MAC sublayer? Also explain its frame format.
27. With the help of diagrams describe Bluetooth architecture.
28. List advantages, disadvantages and applications of Bluetooth.

April 2016
1. Apply bit stuffing on the pattern: 01001111111101111110. [1 M]
2. List any two channelization protocols. [1 M]
3. Explain Pure ALOHA and slotted ALOHA with example. [5 M]
Ans. Refer to Section 2.4.3.1 Point (1).
4. Explain circuit switching in detail. [5 M]
5. Explain Polling "Select" function. [3 M]
6. Which types of services are defined by IEEE 802.11? [1 M]
2.94
7. What is Ethernet? Give its type with a short description. [5 M]

8. Give any four goals of Fast Ethernet. [2 M]
April 2017
1. What is framing? Explain any two framing methods with examples. [4 M]
Ans. Refer to Sections 2.2.3 and 2.3.
2. Explain FDMA in detail. [4 M]
Ans. Refer to Section 2.4.5 Point (1).
3. What are Random access methods? Explain any one mechanism. [4 M]
5. Which standard is used for wireless LAN? [1 M]
Ans. Refer to Section 2.7.
6. What is Bluetooth? Write any two advantages of Bluetooth. [2 M]
Ans. Refer to Sections 2.7.7 and 2.7.7.4.
October 2017
1. State the strategies used to avoid collisions. [1 M]
2. Explain the packet switching with advantages and disadvantages. [5 M]
3. Describe the functions performed by Data Link Layer. [5 M]
Ans. Refer to Sections 2.0 and 2.2.
4. Consider a CDMA scheme with 3 stations having chip sequences [+1 –1 +1 –1],
[+1 +1 –1 –1]. Station 1 sends bit 1. Station 2 sends bit 0. Station 3 is silent.
Show the process of encoding and decoding along with the signals. [5 M]
5. Describe Pure and Slotted ALOHA in brief. [5 M]
6. Why is CSMA/CD not required in full duplex switched Ethernet? [1 M]
2.95
7. State any two applications of wireless LAN. [1 M]

8. Explain in detail 802.3 MAC frame format. [3 M]
9. Write any two applications of Bluetooth technology. [2 M]
April 2018
1. What is the responsibility of a Physical Layer. [1 M]
2. List the cables used with Ethernet LAN. [1 M]
3. A telephone network is an example of a circuit switched network. State
True/False. [1 M]
4. Consider a CDMA scheme with 3 stations having chip sequences [+1 –1 +1 –1],
[+1 +1 –1 –1] and [1+ +1 +1 +1].
Station 1 sends bit 1. Station 2 sends bit 0. Station 3 is silent. Show the process of
encoding and decoding along with the signals. [5 M]
5. Explain the strategies used by CSMA/CA. [4 M]
6. What is Framing? List methods of framing. [2 M]
7. Compare the circuit and packet switching. [4 M]
Ans. Refer to Page No 2.51.
8. Draw BSS with an access point. [1 M]
9. Draw the frame format of Ethernet. [2 M]
October 2018
1. If the bandwidth of the channel is 10 kbps, how long does it take to transmit a
frame of 100000 bits? [1 M]
2.96
2. What is flow control? Why is it needed? [1 M]

3. Calculate the total delay for a frame of size 5 million bits which is sent on a link
with 10 routers, each having queuing time 2 μs and a processing time of 1 μs. The
8
length of the link is 2000 km and speed of light is 2 × 10 m/s in the link. The link
has bandwidth 5 mbps. [5 M]
4. What is framing? Explain any two framing methods with examples. [5 M]
5. Write a note on the Reservation method used in controlled access. [3 M]
6. What is channelization? List three channelization methods and explain any one
method. [5 M]
7. State the minimum and maximum Ethernet frame length. [1 M]
Ans. Refer to 2.6.1.1, Point (3).
8. How many maximum no of computers is used to create Piconet? [1 M]
9. Explain the types of traditional Ethernet. [5 M]
11. Explain the datagram circuit and virtual circuit in detail. [4 M]
12. Explain BSS and its types with diagrams. [4 M]
April 2019
1. If a composite signal is composed of five sine waves of frequencies 100, 300, 500,
700 and 900 Hz. What is the bandwidth of the signal? [1 M]
2. Calculate maximum bit rate using Shannon’s theorem for a channel having
bandwidth 31000 Hz and S/N ratio 20dB. [5 M]
2.97
3. Explain the data link protocols for noiseless channels. [5 M]

4. State the difference between reservation and polling. [3 M]
Ans. Refer to Section 2.4.4 Points (1) and (2).
5. What is channelization? List the methods of channelization. Explain any one
method. [5 M]
6. Give difference between WLAN and Bluetooth. [5 M]
Ans. Refer to Page No. 2.84.
7. Explain 10Base5, 10Base2, 10Base-T, 10Base-F. [4 M]
2.98
CHAPTER
3
Network Layer
Objectives…
To understand Network Layer
To study IPv4 and Addressing
To learn Basic Concepts in IPv6
To understand Congestion Control
To learn Mobile IP and Routing
3.0 INTRODUCTION (April 16)
• Network layer is responsible for host-to-host delivery of the messages (datagrams)

between two hosts.
• The network layer controls the source to destination delivery of data packets across
multiple hops (nodes).
• Network layer is responsible for routing packets from the source host to the
destination host. Network layer also provides mechanisms for congestion control.
• Network layer is also responsible for fragmentation in which large size packets are
broken down into smaller packets/fragments.
Transport Layer
Gives services to
Network Layer
Multicasting
Internetworking Routing
Address
Addressing resolution
Routing
Packetizing Fragmenting
protocols
Receives services from
Data Link Layer
Fig. 3.1: Position of Network Layer
3.1
Computer Networks – I (S.Y.B.Sc. Comp. Science) Network Layer
• When the data packets are routed to remote locations, a logical addressing scheme is
required to differentiate between the source system and the destination system. This is
provided by the network layer.
• One of the main functions of network layer is to provide internetworking
between different networks. It provides a logical connection between different types
of network.
• Fig. 3.1 shows position of network layer in OSI Model.
3.1 NETWORK LAYER SERVICES (Oct. 18)
• The services which are offered by the network layer are as follows:
1. Packetizing:
• The process of encapsulating the data received from upper layers of the network (also
called as payload) in a network layer packet at the source and decapsulating the
payload from the network layer packet at the destination is known as packetizing.
• In other words, one responsibility of the network layer is to carry the data packets
from the source to the destination without changing it or using it.
• The network layer is doing the service of a carrier such as the postal office, is
responsible for delivery of packages from sender to a receiver without changing or
using the content.
• The source host adds a header that contains the source and destination address and
some other relevant information required by the network layer protocol to the
payload received from the upper layer protocol, and delivers the packet to the data
link layer.
• The destination host receives the network layer packet from its data link layer,
decapsulates the packet, and delivers the payload to the corresponding upper layer
protocol.
• The routers in the path are not allowed to change either the source or the destination
address.
• The routers in the path are not allowed to decapsulate the packets they receive unless
they need to be fragmented.
• Packetizing is done by Internet Protocol (IP) that defines its own packet format.
2. Routing and Forwarding:
• Routing and forwarding are two other services offered by the network layer.
• The network layer is responsible for routing the packets from its source to the
destination. With the help of forwarding, data packets are transferred from one place
to another in the network.
3.2
• In a network, there are a number of roots available from the source to the destination.
The network layer has some strategies which find out the best possible route. This
process is referred to as routing.
• There are a number of routing protocols which are used in the routing process and
they should be run to help the routers coordinate with each other and help in
establishing communication throughout the network.
• Forwarding is simply defined as the action applied by each router when a packet
arrives at one of its interfaces.
• Normally, a router uses the decision making table for applying this action sometimes
called the forwarding table and sometimes the routing table.
• When a router receives a packet from one of its attached networks, it needs to forward
the packet to another attached network (unicast routing) or to some attached networks
(in case of multicast routing).
Other Services:
1. Flow Control:
• It regulates the amount of data a source can send without overloading the receiver.
• If the source produces data at a very faster rate than the receiver can consume it, the
receiver will be overloaded with data.
• To control the flow of data, the receiver should send feedback to the sender to inform
the latter that it is overloaded with data.
• There is a lack of flow control in the design of the network layer. It does not directly
provide any flow control.
• The datagrams are sent by the sender when they are ready, without any attention to
the readiness of the receiver.
2. Error Control:
• Error control also in the network layer is the process of detecting and correcting data
packets that have been corrupted or lost during transmission.
• For error control network layer uses ICMP protocol. Internet Control Message Protocol
(ICMP) is a network layer protocol that reports errors and provides information
related to IP packet processing.
• ICMP is used by network devices to send error messages indicating, for example, that a
requested service is not available or that a host is not reachable.
3. Quality of Service:
• The Internet has thrived by providing better quality of service to support applications
like real-time communication of audio and video.
3.3
4. Congestion Control:
• Congestion is a situation in a network layer in which too many datagrams are present
in an area of the Internet.
• Congestion occurs when the number of datagrams sent by source is beyond the
capacity of network or routers.
• If congestion continues, sometimes a situation may arrive where the system collapses
and no datagrams are delivered.
• Although congestion control is indirectly implemented in the network layer, there is
still a lack of congestion control in the network layer.
5. Security:
• The network layer was designed with no security provision. The network layer uses a
virtual level IPSec for security.
3.2 CONGESTION CONTROL (OPEN AND CLOSED LOOP)

• Congestion control refers to the techniques used to control or prevent congestion.
Basic Concept of Congestion: (Oct. 18, April 19)
• Congestion in a network may occur if the load on the network (the number of packets
sent to the network) is greater than the capacity of the network (the number of packets
a network can handle).
• Congestion control refers to the mechanisms and techniques to control the congestion
and keep the load on the network below the capacity.
Fig. 3.2: When too Many Packets are Sent, Congestion Occurs and Performance Degrades
3.4
• At very high traffic, performance collapses completely and almost no packets are
delivered. Buffers get full, so packets are discarded leading to more retransmissions
and less packets delivered to their destinations.
• Congestion thus tends to feed upon itself and become worse, leading to collapse of the
system.
• Congestion can occur because of several reasons:
1. If there is insufficient memory at the router to hold the packets, congestion occurs
and packets are lost.
2. Slow processors can also cause congestion. If the router’s processor is slow, queues
can build up, even though there is excess line capacity, congestion can occur.
3. Similarly, low bandwidth lines can also cause congestion.
Basic Concept of Congestion Control:
• Congestion control refers to techniques and mechanisms that can either prevent
congestion, before it happens, or remove congestion, after it has happened.
• Congestion control mechanisms are divided into two categories, one category prevents
the congestion from happening (open loop congestion control) and the other category
removes (closed loop congestion control) congestion after it has taken place.
Congestion
control
Open loop Closed loop
Fig. 3.3: Mechanisms for Congestion Control
1. Open Loop Congestion Control: (Oct. 17)

• Open loop congestion control policies are applied to prevent congestion before it
happens.
• The congestion control is handled either by the source or the destination.
• Policies adopted by open loop congestion control are explained below:
(i) Retransmission Policy: It is the policy in which retransmission of the packets
are taken care. If the sender feels that a sent packet is lost or corrupted, the
packet needs to be retransmitted. This transmission may increase the congestion
in the network. To prevent congestion, retransmission timers must be designed
to prevent congestion and also able to optimize efficiency.
(ii) Window Policy: The type of window at the sender side may also affect the
congestion. Several packets in the Go-Back-N window are present, although some
3.5
packets may be received successfully at the receiver side. This duplication may
increase the congestion in the network and make it worse. Therefore, a Selective
repeat window should be adopted as it sends the specific packet that may have
been lost.
(iii) Discarding Policy: A good discarding policy adopted by the routers is that the
routers may prevent congestion and at the same time partially discards the
corrupted or less sensitive package and also be able to maintain the quality of a
message. In case of audio file transmission, routers can discard less sensitive
packets to prevent congestion and also maintain the quality of the audio file.
(iv) Acknowledgment Policy: Since acknowledgement is also part of the load in the
network, the acknowledgment policy imposed by the receiver may also affect
congestion. Several approaches can be used to prevent congestion related to
acknowledgment. The receiver should send acknowledgement for N packets
rather than sending acknowledgement for a single packet. The receiver should
send an acknowledgment only if it has to send a packet or a timer expires.
(v) Admission Policy: In admission policy a mechanism should be used to prevent
congestion. Switches in a flow should first check the resource requirement of a
network flow before transmitting it further. If there is a chance of a congestion
or there is a congestion in the network, the router should deny establishing a
virtual network connection to prevent further congestion.
• All the above policies are adopted to prevent congestion before it happens in the
network.
2. Closed Loop Congestion Control: (Oct. 17)
• Closed loop congestion control technique is used to treat or alleviate congestion after it
happens. Closed loop solutions are based on feedback loops.
• Closed loop congestion control has following three parts:
(i) Monitor the system to detect when and where congestion occurs.
(ii) Pass this information to places where action can be taken.
(iii) Adjust system operations to correct the problem.
• Policies adopted by open loop congestion control are explained below:
(i) Backpressure: Backpressure is a technique in which a congested node stops
receiving packets from upstream nodes. This may cause the upstream node or
nodes to become congested and rejects receiving data from above nodes.
Backpressure is a node-to-node congestion control technique that propagates in
the opposite direction of data flow. The backpressure technique can be applied
3.6
only to virtual circuits where each node has information of its above upstream
node.
rd nd
In the Fig. 3.4, the 3 node is congested and stops receiving packets as a result 2
st
node may get congested due to slowing down of the output data flow. Similarly, 1
node may get congested and inform the source to slow down.
Fig. 3.4
(ii) Choke Packet Technique: Choke packet technique is applicable to both virtual
networks as well as datagram subnets. A choke packet is a packet sent by a node
to the source to inform it of congestion. Each router monitors its resources and
the utilization at each of its output lines. whenever the resource utilization
exceeds the threshold value which is set by the administrator, the router directly
sends a choke packet to the source giving it a feedback to reduce the traffic. The
intermediate nodes through which the packets have traveled are not warned
about congestion. Fig. 3.5 shows the idea of a choke packet.
Fig. 3.5
(iii) Implicit Signaling: In implicit signaling, there is no communication between the
congested nodes and the source. The source guesses that there is congestion in a
network. For example, the when the sender sends several packets and there is no
acknowledgement for a while, one assumption is that there is a congestion.
(iv) Explicit Signaling: In explicit signaling, if a node experiences congestion it can
explicitly send a packet to the source or destination to inform about congestion.
The difference between choke packet and explicit signaling is that the signal is
included in the packets that carry data rather than creating different packets as in
case of choke packet technique. Explicit signaling can occur in either forward or
backward direction.
3.7
o Forward Signaling: In forward signaling a signal is sent in the direction of

the congestion. The destination is warned about congestion. The receiver in
this case adopts policies to prevent further congestion.
o Backward Signaling: In backward signaling a signal is sent in the opposite
direction of the congestion. The source is warned about congestion and it
needs to slow down.
3.3 IPv4 ADDRESSING

• The Internet Protocol version 4 (IPv4), is responsible for packetizing, forwarding and
delivery of a packet at the network layer.
• Internet Protocol (IP) is a set of rules that dictate how data should be delivered over
the Internet.
• Networks using the TCP/IP protocol route messages based on the IP address of the
destination.
• IP addressing is the method used to identify hosts and network devices in a network.
• The Internet Protocol Address (IP Address) is a unique number assigned to every
computing device, such as personal computers, tablets, and smartphones used to
identify itself and communicate with other devices in the IP network. Any device
connected to the IP network must have a unique IP address within the network.
• Following are the two versions of the Internet Protocol (IP) are in common uses in the
Internet today:
1. An Internet Protocol version 4 (IPv4) address is a 32-bits address that uniquely
and universally defines the connection of a host or a router to the Internet.
2. The growth of the Internet and the depletion of available IPv4 addresses, a new
version of IP, Internet Protocol version 6 (IPv6), using 128-bits for the IP address,
was developed in 1995.
• In this section, we study IP addressing in detail with IPv4 addressing. The IPv4
addresses are universal in the sense that the addressing system must be accepted by
any host that wants to be connected to the internet.
Basic Concept of IP Address/IPv4 Address: (April 16, 17, 18, 19; Oct. 17)
• An IP address is an address used to uniquely identify a device on an IP network. The
identifier used in the TCP/IP protocol suite to identify the connection of each device to
the Internet called as the Internet Protocol address (IP address).
• An IP address is a numerical (32-bits) representation that uniquely identifies a specific
interface on the network.
• An IP address or logical address is an identifier assigned to each computer and other
device (e.g., printer, router etc.) connected to a TCP/IP network that is used to locate
and identify the node in communications with other nodes on the network.
3.8
• An IPv4 address is a 32-bit address that uniquely and universally defines the
connection of a device such as a computer or a router to the Internet. Fig. 3.6 shows an
IP address format.
• An IP address consists of two parts namely, a Network ID, which specifies the network
on which a host resides, and a Host ID, which identifies the host within that network.
Fig. 3.6: Parts/Format of an IP Address Fig. 3.7: Example of an IP Address
3.3.1 Address Space

• IPv4 protocol defines addresses that have an address space. An address space is the
total number of addresses used by the protocol.
• IPv4 uses 32-bits addresses, which means the address space is 232 or 4,294,967, 296
(more than 4 billion), this means if there were no restrictions, more than 4 billion
devices could be connected to the Internet.
• There are three common
notations to show an IPv4
address namely, binary
notation (base 2), dotted-
decimal notation (base 256),
and hexadecimal notation
(base 16).
• Fig. 3.8 shows all the three
notations for IPv4 address. Fig. 3.8: Three Notations of IPv4 Addressing
• The notations for IP address are explained below:

1. Binary Notation (Base 2): Binary notation is the format that systems on the
network use to process the address. In binary notation the IPv4 address is
displayed as 32 bits. Each octet is often referred to as a byte. So it is common to
hear an IPv4 address referred to as a 32-bit address or a 4-byte address. An
example of binary notation is 01110101.10010101.00011101.11101010.
2. Hexadecimal Notation (Base 16): We sometimes see an IPv4 address in
hexadecimal notation. Each hexadecimal digit is equivalent to four bits. This
means that a 32-bits address has 8 hexadecimal digits. This notation is often used
in network programming. An example of hexadecimal notation of an IPv4 address
is C0.A8.01.64.
3.9
3. Dotted-decimal Notation (Base 256): To make the IPv4 address more compact
and easier to read, Internet addresses are usually written in decimal form with a
decimal point (dot) separating the bytes. Dotted-decimal notation is the format that
is typically used for displaying the IP address in a human-readable format. An
example of dotted-decimal notation is 192.168.1.100.
Examples:
Example 1: Change the following IPv4 addresses from binary notation to dotted-
decimal notation.
(i) 10000001 00001011 00001011 11101111
(ii) 11000001 10000011 00011011 11111111
(iii) 11100111 11011011 10001011 01101111
(iv) 11111001 10011011 11111011 00001111
Solution:
We replace each group of 8 bits with its equivalent decimal number and add dots
for separation.
(i) 129.11.11.239
(ii) 193.131.27.255
(iii) 231.219.139.111
(iv) 249.155.251.15
Example 2: Change the following IPv4 addresses from dotted-decimal notation to
binary notation.
(i) 111.56.45.78
(ii) 221.34.7.82
(iii) 241.8.56.12
(iv) 75.45.34.78
Solution:
We replace each decimal number with its binary equivalent.
(i) 01101111 00111000 00101101 01001110
(ii) 11011101 00100010 00000111 01010010
(iii) 11110001 00001000 00111000 00001100
(iv) 01001011 00101101 00100010 01001110
Example 3: Change the following IPv4 addresses from binary notation to hexadecimal
notation.
(i) 10000001 00001011 00001011 11101111
3.10
(ii) 11000001 10000011 00011011 11111111

Solution:
We replace each group of 4 bits with its hexadecimal equivalent. Note that
hexadecimal notation normally has no added spaces or dots; however, 0X (or 0x) is
added at the beginning or the subscript 16 at the end to show that the number is in
hexadecimal.
(i) 0X810B0BEF or 810B0BEF16
(ii) 0XC1831BFF or C1831BFF16
Example 4: Convert the following IPv4 addresses from binary notation to decimal
notation.
(i) 01110101 10010101 00011101 00000100, and
(ii) 10000001 00001011 00001001 11101111.
Solution: (i) 117.149.29.4
(ii) 129.11.9.239.
Example 5: Convert the following IPv4 addresses from decimal notation to binary
notation if it is in correct form.
(i) 128.29.4.31
(ii) 221.34.7.82
(iii) 221.36.3.4.5
(iv) 129.300.4.10.
Solution: (i)10000000 00011101 00000100 00011111.
(ii) 11011101 00100010 00000111 01010010.
(iii) There is an error, no more than 4 numbers in an IPv4 address.
(iv) There is an error, Each number should be less than or equal to 255 (300 is out of
range).
3.3.2 Classful Addressing (April 16)
• Internet Protocol (IP) hierarchy contains several classes of IP Addresses to be used

efficiently in various situations as per the requirement of hosts per network.
• IPv4 addressing uses the concept of classes. This architecture is called classful
addressing.
• In classful addressing, the address space is divided into five classes namely A, B, C, D
and E. Each class in classful addressing, occupies some part of the whole address
space.
3.11
• In class A, the network length is 8 bits, but since the first bit, which is 0, defines the
class, we can have only seven bits as the network identifier. This means there are only
7
2 = 128 networks in the world that can have a class A address.
• In class B, the network length is 16 bits, but since the first two bits, which are (10)2,
define the class, we can have only 14 bits as the network identifier. This means there
14
are only 2 = 16384 networks in the world that can have a class B address.
• All addresses that start with (110)2, belong to class C. In class C, the network length is
24 bits, but since three bits define the class, we can have only 21 bits as the network
21
identifier. This means there are 2 = 2,097,152 networks in the world that can have a
class C address.
• Class D (11102 is not divided into NetworkID and HostID. It is used for multicast
addresses. All addresses that start with 1111 in binary belong to class E. As in Class D,
Class E is not divided into NetworkID and HostID and is used as reserve.
Fig. 3.9: Classful Addressing

Examples:
Example 1: Find the class of each address:
(i) 00000001 00001011 00001011 11101111
(ii) 11000001 10000011 00011011 11111111
(iii) 10100111 11011011 10001011 01101111
(iv) 11110011 10011011 11111011 00001111
Solution:
(i) The first bit is 0. This is a class A address.
3.12
(ii) The first 2 bits are 1; the third bit is 0. This is a class C address.
(iii) The first bit is 1; the second bit is 0. This is a class B address.
(iv) The first 4 bits are 1s. This is a class E address.
Example 2: Find the class of each address:
(i) 227.12.14.87
(ii) 193.14.56.22
(iii) 14.23.120.8
(iv) 252.5.15.111
Solution:
(i) The first byte is 227 (between 224 and 239); the class is D.
(ii) The first byte is 193 (between 192 and 223); the class is C.
(iii) The first byte is 14 (between 0 and 127); the class is A.
(iv) The first byte is 252 (between 240 and 255); the class is E.
Basic Concept of Masking:
• A mask used to determine what subnet an IP address belongs to. A process that
extracts the address of the physical network from an IP address is called Masking. If
we do the subnetting, then masking extracts the subnetwork address from an IP
address.
• It may at first seem to be odd that IP address classes are assigned in this way. After all,
there are not any private networks that have 16 million hosts on them, so it makes
little sense even to have Class A addresses. However, it's possible to subdivide IP
addresses even further by creating subnets on them.
• A subnet is simply a subdivision of a network address that can be used to represent
one LAN on an internetwork or the network of one of an ISP's clients.
• Thus, a large ISP might have a Class A address registered to it and it might farm out
pieces of the address to its clients in the form of subnets.
• In many cases, a large ISP's clients are smaller ISPs, which in turn supply addresses to
their own clients.
• A subnet allows the flow of network traffic between hosts to be segregated based on a
network configuration.
• A subnet mask (or number) is used to determine the number of bits used for the
subnet and host portions of the address.
• The mask is a 32-bit value that uses one-bits for the network and subnet portions and
zero-bits for the host portion.
3.13
Subnet Masks:
• IP networks can be divided into smaller networks called subnetworks (or subnets).
The subnets are created through the use of subnet masks.
• The subset mask identifies which hits in the IP address are to be used to represent the
network subnet portion of an IP address.
• The network mask is used when a network is not subnetted.
• When we divide a network
into several subnetworks,
we need to create a
subnetwork mask (or subnet
mask) for each subnetwork.
• Fig. 3.10 shows a
subnetwork has sub netid Fig. 3.10: Network Mask and Subnetwork Mask
and hostid.
• Subnetting provides the following advantages:
o Network Traffic Isolation: There is less network traffic on each subnet.
o Simplified Administration: Networks may be managed independently.
o Improved Security: Subnets can isolate internal networks so they are not visible
from external networks.
• A 14-bits subnet mask on a class B network only allows 2 node addresses for WAN
links. A routing algorithm like OSPF (Open Shortest Path First) must be used for this
approach.
• These protocols allow the Variable Length Subnet Masks (VLSM). RIP (Routing
Information Protocol) and IGRP (Interior Gateway Routing Protocol) don't support this.
• Subnet mask information must be transmitted on the update packets for dynamic
routing protocols for this to work.
• The router subnet mask is different from the WAN interface subnet mask. One
network ID is required by each of:
o Subnet,
o WAN connection.
• One host ID is required by each of:
o Each NIC on each host.
o Each router interface.
• Types of subnet masks:
o Default: Fits into a Class A, B, or C network category.
o Custom: Used to break a default network such as a Class A, B or C network into
subnets.
3.14
Fig. 3.11: Masking Concept
Fig. 3.12: AND Operation
Default Masks
Class Mask in Binary Mask in Dotted-decimal
A 11111111 00000000 00000000 00000000 255.0.0.0
B 11111111 11111111 00000000 00000000 255.255.0.0
C 11111111 11111111 11111111 00000000 255.255.255.0
(a) Without Subnetting (b) With Subnetting
Fig. 3.13: Default Mask and Subnet Mask
Fig. 3.14: Host Communication on a Local Network
• A subnet is defined by applying a bitmask, the subnet mask, to the IP address. If a bit is
on the mask, the equivalent bit in the address is interpreted as a network bit.
3.15
• If the bit in the mask is off, the bit belongs to the host part of the address. The subnet is
only known locally. To the rest of the Internet, the address is still interpreted as a
standard IP address.
Fig. 3.15: Host Communication with Subnetting
Examples: Find the subnetwork address for the following.

Sr. No. IP Address Mask
1. 141.181.14.16 255.255.225.0
2. 200.34.22.156 255.255.255.240
3. 125.35.12.57 255.255.0.0
Solution:
1. 141.181.14.16 IP Address
255.255.224.0 Mask
141.181.0.0 Subnetwork Address
2. 200.34.22.156 IP Address
255.255.255.240 Mask
3. 125.35.12.57 IP Address
255.255.0.0 Mask
3.3.3 Subnetting (April 18)

• If an organization was granted a large block in class A or B, it could divide the
addresses into several continuous groups and assign each group to smaller networks
(called subnets).
• The subnets are created through the use of subnet masks. The subnet mask identifies
which bits in the IP address are to be used to represent the network/subnet portion of
an IP address.
• Subnetting is defined as, “the process of dividing (partitioning) a network into several
smaller networks (subnets)”.
• Subnetting is a method for partitioning/dividing a classful IP network into smaller
subnetworks (subnets).
3.16
• The process of subnetting involves dividing a network up into smaller networks called
subnets or sub networks.
• A subnet is a logical partition of an IP network into multiple, smaller network
segments. Each of these subnets has its own specific address.
• To create these additional sub networks, we use a subnet mask. The subnet mask
simply determines which portion of the IP address belongs to the host.
• The subnet address is created by dividing the host address into network address and
host address.
• To subnet a network, extend the natural mask using some of the bits from the host ID
portion of the address to create a subnetwork ID.
• For example, given a Class C network of 204.15.5.0 which has a natural mask of
255.255.255.0. We can create subnets in the following manner:
204.15.5.0 11001100.00001111.00000101.00000000
255.255.255.224 11111111.11111111.11111111.11100000
---------------------------------(sub)---------------
• By extending the mask to be 255.255.255.224, we have taken three bits (seen above as
"sub") from the original host portion of the address and used them to make subnets.
With these three bits, it is possible to create eight subnets.
• With the remaining five host ID bits. Each subnet can have up to 32 host addresses, 30
of which can actually be assigned to a device since host ids of zeros or all ones are not
allowed (it is very important to remember this).
• So, with this in mind, the following subnets have been created:
204.15.5.0 255.255.255.224 host address range 1 to 30
• Using the network subnetting scheme above, which allows for eight subnets, the
network might appear as shown in Fig. 3.16.
Fig. 3.16
3.17
• Fig. 3.17 shows a network using class B addresses before subnetting. In this example
we have just one network with almost 216 hosts.
• The whole network is connected, through one single connection, to one of the routers
in the Internet.
Fig. 3.17
• Fig. 3.18 shows the same network in Fig. 3.17 after subnetting. The whole network is
still connected to the Internet through the same router.
• However, the network has used a private router to divide the network into four
subnetworks.
Fig. 3.18
3.18
Examples:
Example 1: Assume a company has three offices Central, East, and West. The Central
office is connected to the East and West offices via private, point-to-point WAN lines. The
company is granted a block of 64 addresses with the beginning address 70.12.100.128/26.
The management has decided to allocate 32 addresses for the Central office and divides
the rest of addresses between the two other offices.
1. The number of addresses are assigned as follows:
Central office Nc = 32 East office Ne = 16 West office Nw = 16
2. We can find the prefix length for each subnetwork:
nc = n + log2(64/32) = 27 ne = n + log2 (64/16) = 28 nw = n + log2(64/16) = 28
Solution:
Fig. 3.19 shows the configuration designed by the management. The Central office uses
addresses 70.12.100.128/27 to 70.12.100.159/27. The company has used three of these
addresses for the routers and has reserved the last address in the subblock. The East office
uses the addresses 70.12.100.160/28 to 70.12.100.175/28. One of these addresses is used for
the router and the company has reserved the last address in the subblock. The West office
uses the addresses 70.12.100.160/28 to 70.12.100.175/28. One of these addresses is used for
the router and the company has reserved the last address in the subblock. The company
uses no address for the point-to-point connections in WANs.
Fig. 3.19
Example 2: An organization is granted the block 130.34.12.64/26. The organization

needs four subnetworks, each with an equal number of hosts. Design the subnetworks and
find the information about each network.
3.19
Solution:
32–26
The number of addresses for the whole network can be found as N = 2 = 64. The
first address in the network is 130.34.12.64/26 and the last address is 130.34.12.127/26. We
now design the subnetworks:
1. We grant 16 addresses for each subnetwork to meet the first requirement (64/16 is a
power of 2).
2. The subnetwork mask for each subnetwork is:
n1 = n2 = n3 = n4 = n + log2 (N/Ni) = 26 + log24 = 28
3. We grant 16 addresses to each subnet starting from the first available address.
Fig. 3.20 shows the subblock for each subnet. Note that the starting address in each
subnetwork is divisible by the number of addresses in the subnetwork.
(a) Original Block
(b) Sub Blocks

Fig. 3.20
3.3.4 Supernetting
• When most of the class A and class B addresses were depleted, there was a huge
demand for midsize blocks.
• The size (only 256) of class C was not sufficient. Even a midsize organization needs
more than 256 addresses. Supernetting is the solution for this problem.
• In supernetting, an organization can combine several class C blocks to create a larger
range of addresses. Several networks are combined to create a supernetworks or a
supernet.
• Supernetting, also called Classless Inter-Domain Routing (CIDR), is a way to aggregate
multiple Internet addresses of the same class.
3.20
• Classful means that the IP addresses and subnets are within the same network. The
problem with classful addressing is that there is a lot of unused IP address space.
• For example, a class A IP network has more than 16 million possible host addresses. A
Class B network has more than 65,000 host addresses, but the fact is that only a limited
number of Class A and B address space has been allocated for Internet use.
• However, the size of a class C block with a maximum number of 256 addresses may
not satisfy the needs of an organization. Even a mid-size organization may need more
addresses.
• One solution is supernetting. In supernetting, an organization can combine several
class C blocks to create a larger range of addresses.
• In other words, several networks are combined to create a supernetwork. By doing
this, an organization can apply for a set of class C blocks instead of just one.
• For example, an organization that needs 1000 addresses can be granted four class C
blocks.
• The organization can then use these addresses in one supernetwork as shown in
Fig. 3.21. When we group two or more classful networks together, they are called
supernets.
• The technique supernetting was proposed in 1992 to eliminate the class boundaries
and to make available the unused IP address space.
• Supernetting allows multiple networks to be specified by one subnet mask. In other
words, the class boundary could be overcome.
Fig. 3.21: A Supernetwork
3.21
• Supernetting required a simpler way to indicate the subnet mask. The technique
developed is called Classless Inter-Domain Routing (CIDR). CIDR notation specifies the
number of bits set to a 1 that make up the subnet mask.
• For example, the Class C size subnet mask 255.255.255.0 is listed in CIDR notation as
/24. This indicates the 24 bits are set to a 1. A Class B size subnet is written as /16, and a
Class A subnet is written as /8.
• CIDR can also be used to represent subnets that identify only part of the octet bits in an
IP address. For example, a subnet mask of 255.255.192.0 is written in CIDR as /18.
• The /18 comes from the 18 bits that are set to a 1 as shown below:
255 255 192 0
11111111 11111111 11000000 00000000
• CIDR notation truncates the subnet mask to what is known as “slash” notation. In this
example, the network would be identified as 131.107.0.0/16.
• The “/16” value refers to the fact that the first 16 bits in the subnet mask are all set to
values of binary 1.
Fig. 3.22
Advantages of Supernetting:
1. Control and reduce network traffic.
2. Helpful to solve the problem of lacking IP addresses.
3. Minimizes the routing table.
Disadvantages of Supernetting:
1. It cannot cover different areas of the network when combined.
2. All the networks should be in the same class and all IP should be contiguous.
3.3.5 Classless Addressing

• The fast growth of the Internet led to the near depletion of the available addresses.
We have run out of class A and B addresses, and a class C addresses is too small for
most midsize organizations.
• To overcome the problem of address depletion and give more organizations access to
the Internet, classless addressing was designed and implemented.
• In classless addressing, there are no classes, but the addresses are still granted in
blocks.
• In 1996, the Internet authorities announced a new architecture called classless
addressing. In classless addressing, variable-length blocks are used that belong to no
3.22
classes. We can have a block of 1 address, 2 addresses, 4 addresses, 128 addresses, and
so on.
• In classless addressing, the whole address space is divided into variable length blocks.
0 1 2 32
Theoretically, we can have a block of 2 , 2 , 2 , ..… 2 addresses. The only restriction, is
that the number of addresses in a block needs to be a power of 2. An organization can
be granted one block of addresses.
• Fig. 3.23 shows the division of the whole address space into non-overlapping blocks.
• In the classful addressing, routers could just assume the class of an address based on
the network ID. In the classless addressing, subnet mask information must always be
provided when routers exchange information with each other.
• Some routing protocols, such as the Border Gateway Protocol (BGPv4) and OSPF,
support classless addressing.
Fig. 3.23: Variable Length Blocks in Classless Addressing
• In classless addressing, when an entity, small or large, needs to be connected to the

Internet, it is granted a block or range of addresses.
• The size of the block (the number of addresses) varies based on the nature and size of
the entry.
• For example, a household may be given only two addresses; a large organization may
be given thousands of addresses. An ISP, may be given thousands or hundreds of
thousands based on the number of customers it may serve.
• In classless addressing variable-length blocks are assigned that belong to no class. In
this architecture, the entire address space (232 addresses) is divided into blocks of
different sizes.
• All IP addresses have a network and host portion. Classless addressing uses a variable
number of bits for the host portions of the address.
Decimal 192 160 20 48
Binary 11000000 10100000 00010100 00110000
28 bits network 4 bits host
• Classless addressing treats the IP address as a 32-bits stream of ones and zeros, where
the boundary between network and host portion can fall anywhere between bit 0 and
bit 31.
3.23
• The network portion of an IP address is determined by how many 1's are in the subnet
mask. Again, this can be a variable number of bits, and although it can fall on an octet
boundary, it does not necessarily need to.
3.3.6 Network Address Translation (NAT)

• The number of home users and small businesses who want to use the Internet is day-
by-day increasing. ISP assigns a block of addresses to its user. An address is assigned to
a user when it is needed.
• Additionally, users require more addresses for their small networks. With the
shortage of addresses, this is a serious problem. A solution to this problem is Network
Address Translation (NAT).
• A technology that can provide the mapping between the private and universal
addresses, and at the same time support Virtual Private Networks (VPNs) is Network
Address Translation (NAT).
• The NAT technology allows a site to use a set of private addresses for internal
communication and a set of global Internet addresses (at least one) for communication
with the rest of the world.
• The site must have only one connection to the global Internet through a NAT-capable
router that runs NAT software.
• Fig. 3.24 shows a simple implementation of NAT. NAT is a method of remapping one IP
address space into another by modifying network address information in the IP
header of packets while they are in transit across a traffic routing device.
• The technique was originally used as a shortcut to avoid the need to readdress every
host when a network was moved.
Fig. 3.24
• As the Fig. 3.24 shows, the private network uses private addresses.
• The router that connects the network to the global address uses one private address
and one global address. The private network is invisible to the rest of the Internet; the
rest of the Internet sees only the NAT router with the address 200.24.5.8.
3.24
Concept of Address Translation:

• NAT is a process in which one or more local IP addresses are translated into one or
more Global IP addresses and vice versa in order to provide Internet access to the local
hosts.
• NAT translates the IP addresses of computers in a local network to a single IP address.
This address is often used by the router that connects the computers to the Internet.
• All of the outgoing packets go through the NAT router, which replaces the source
address in the packet with the global NAT address.
• All incoming packets also pass through the NAT router, which replaces the destination
address in the packet (the NAT router global address) with the appropriate private
address.
• Fig. 3.25 shows an example of address translation.
Fig. 3.25: Address Translation

Translation Table:
• The reader may have noticed that translating the source addresses for an outgoing
packet is straightforward. But how does the NAT router know the destination address
for a packet coming from the Internet? There may be tens or hundreds of private IP
addresses, each belonging to one specific host. The problem is solved if the NAT router
has a translation table.
• The NAT table is what allows devices on a private network to access a public network,
such as the internet.
• A NAT table is a table of network address translations, where each row in the table is
basically a mapping from one private address to one public address.
Using One IP Address:
• In its simplest form, a translation table has only two columns namely the private
address and the external address (destination address of the packet).
• When the router translates the source address of the outgoing packet, it also makes
note of the destination address-where the packet is going.
3.25
• When the response comes back from the destination, the router uses the source
address of the packet (as the external address) to find the private address of the
packet.
• Fig. 3.26 shows the idea.
Fig. 3.26
Using a Pool of IP Addresses:
• The use of only one global address by the NAT router allows only one private-network
host to access a given external host. To remove this restriction, the NAT router can use
a pool of global addresses.
• For example, instead of using only one global address (200.24.5.8), the NAT router can
use four addresses (200.24.5.8, 200.24.5.9, 200.24.5.10, and 200.24.5.11).
• In this case, four private-network hosts can communicate with the same external host
at the same time because each pair of addresses defines a separate connection.
Using Both IP Addresses and Port Addresses:
• To allow a many-to-many relationship between private-network hosts and external
server programs, more information in the translation table is required.
• For example, suppose two hosts with addresses 172.18.3.1 and 172.18.3.2 inside a
private network need to access the HTTP server on an external host 25.8.3.2.
3.26
• If the translation table has five columns, instead of two, that include the source and
destination port numbers of the transport layer protocol, the ambiguity is eliminated.
• Table 3.1 shows an example of such a table.
Table 3.1: Five-column Translation Table
Private Private External External Transport
Address Port Address Port Protocol
172.18.3.1 1400 25.8.3.2 80 TCP

172.18.3.2 1401 25.8.3.2 80 TCP
... ... ... ... ...
Advantages of NAT:
1. NAT solves IP overlapping issues.
2. NAT hides internal IP structure from the external world.
3. NAT allows us to connect with any network without changing IP address.
4. NAT allows connecting multiple computers with the internet through the single
public IP address.
Disadvantages of NAT:
1. NAT adds additional delay in the network.
2. Several applications are not compatible with NAT.
3. End to end IP traceability will not work with NAT.
4. NAT hides the actual end device.
3.4 FORWARDING OF IP PACKETS

• Forwarding means to place the packet in its route to its destination. Since, the Internet
today is made of a combination of links (networks), forwarding means to deliver the
packet to the next hop (which can be the final destination or the intermediate
connecting device).
• Although the IP protocol was originally designed as a connection less protocol, today
the tendency is to change it to connection-oriented protocol.
• When IP is used as a connection less protocol, forwarding is based on the destination
address of the IP datagram when the IP is used as a connection oriented protocol,
forwarding is based on the label attached to an IP datagram.
3.4.1 Forwarding Based on Destination Address

• Forwarding IP packets based on the destination address is a traditional approach,
which is prevalent today.
3.27
• In this case, forwarding requires a host or a router to have a forwarding table. When a
host has a packet to send or when a router has received a packet to be forwarded, it
looks at this table to find the next hop to deliver the packet to.
• In classless addressing, the whole address space is one entity; there are no classes. This
means that forwarding requires one row of information for each block involved.
• The table needs to be searched based on the network address (first address in the
block). Unfortunately, the destination address in the packet gives no clue about the
network address.
• To solve the problem, we need to include the mask (/n) in the table. In other words, a
classless forwarding table needs to include four pieces of information: the mask, the
network address, the interface number, and the IP address of the next router (needed
to find the link-layer address of the next hop).
• However, we often see in the literature that the first two pieces are combined. For
example, if n is 26 and the network address is 180.70.65.192, then one can combine the
two as one piece of information: 180.70.65.192/26.
• Fig. 3.27 shows a simple forwarding module and forwarding table for a router with
only three interfaces.
Fig. 3.27: Simplified Forwarding Module in Classless Address
• The job of the forwarding module is to search the table, row by row. In each row, the n
leftmost bits of the destination address (prefix) are kept and the rest of the bits (suffix)
are set to 0s.
• If the resulting address (which we call the network address), matches with the address
in the first column, the information in the next two columns is extracted; otherwise
the search continues.
• Normally, the last row has a default value in the first column (not shown in Fig. 3.27),
which indicates all destination addresses that did not match the previous rows.
3.28
• Sometimes, the literature explicitly shows the value of the n leftmost bits that should
be matched with the n leftmost bits of the destination address.
• The concept is the same, but the presentation is different. For example, instead of
giving the address-mask combination of 180.70.65.192/26, we can give the value of the
26 leftmost bits as shown below:
10110100 01000110 01000001 11
• Note that we still need to use an algorithm to find the prefix and compare it with the
bit pattern.
• In other words, the algorithm is still needed, but the presentation is different. We use
this format in our forwarding tables in the exercises when we use smaller address
spaces just for practice.
• For example, make a forwarding table for router R1 using the configuration in
Fig. 3.28.
Fig. 3.28
Solution: Table 3.2 shows the corresponding table.
Table 3.2: Forwarding table for router R1 in Fig. 3.28
Network Address/Mask Next Hop Interface
180.70.65.192/26 - m2
180.70.65.128/25 - m0
201.4.22.0/24 - m3
201.4.16.0/22 - m1
Default 180.70.65.200 m2
3.29
• Fig. 3.29 showed a simple example of searching in a forwarding table using the longest
mask algorithm. Although there are some more efficient algorithms today, the
principle is the same.
• When the forwarding algorithm gets the destination address of the packet, it needs to
delve into the mask column.
• For each entry, it needs to apply the mask to find the destination network address. It
then needs to check the network addresses in the table until it finds the match.
• The router then extracts the next-hop address and the interface number to be
delivered to the data-link layer.
Fig. 3.29: Forwarding Based on Destination Address
3.4.2 Forwarding Based on Label

• In the 1980s, an effort started to somehow change IP to behave like a connection
oriented protocol in which the routing is replaced by switching.
• In a connectionless network (datagram approach), a router forwards a packet based
on the destination address in the header of the packet.
• On the other hand, in a connection-oriented network (virtual-circuit approach), a
switch forwards a packet based on the label attached to the packet.
• Routing is normally based on searching the contents on a table; switching can be done
by accessing a table using an index. In other words, routing involves searching;
switching involves accessing.
3.30
• Fig. 3.30 shows a simple example of using a label to access a switching table. Since, the
labels are used as the index to the table, finding the information in the table is
immediate.
Fig. 3.30: Forwarding Based on Label

• During the 1980s several vendors created routers that implement switching
technology.
• In standard Multiple Protocol Label Switching (MLPS) conventional routers in the
Internet can be replaced by MPLS routers, which can behave like a router and a
switch.
• When behaving like a router, MPLS can forward the packet based on the destination
address; when behaving like a switch, it can forward a packet based on the label.
3.5 NETWORK LAYER PROTOCOLS

• Fig. 3.31 shows the positions of the network layer protocols in the TCP/IP protocol
suite.
• In network layer, Internet Protocol version 4 (IPv4), is responsible for forwarding,
packetizing, and delivery of a packet.
• The Internet Control Message Protocol version 4 (ICMPv4) helps IPv4 to handle
some errors that may occur in the network-layer delivery.
• The Internet Group Management Protocol (IGMP) is used to help IPv4 in
multicasting.
3.31
• The Address Resolution Protocol (ARP) is used to glue the network and data-link
layers in mapping network-layer addresses to link-layer addresses.
Fig. 3.31: Positions of Network Layer Protocols in TCP/IP Protocol Suite
3.5.1 Internet Protocol (IP)

• The Internet Protocol (IP) is a protocol or set of rules, for routing and addressing
packets of data so that they can travel across networks and arrive at the correct
destination.
• IP is an unreliable and connection less datagram protocol. Internet Protocol works at
the network layer.
• Internet Protocol (IP) has the responsibility of identifying hosts based upon their
logical addresses and to route data among them over the underlying network.
• IP provides a mechanism to uniquely identify hosts by an IP addressing scheme. IP
uses best effort delivery, i.e. it does not guarantee that packets would be delivered to
the destined host, but it will do its best to reach the destination.
• There are two versions of IP that currently coexist in the Internet namely, IP version 4
(IPv4) and IP version 6 (IPv6).
• Internet Protocol version 4 (IPv4) is the fourth version of the Internet Protocol (IP).
• IPv4 is one of the core protocols of standards-based internetworking methods in
the Internet and other packet-switched networks. Internet Protocol version 4 uses 32-
bit logical addresses.
• Internet Protocol version 6 (IPv6) is the most recent version of the Internet
Protocol (IP), the communications protocol that provides an identification and location
system for computers on networks and routes traffic across the Internet. Internet
Protocol version 6 uses a 128-bit logical address.
3.32
• In this section we will study Internet Protocol version 4 (IPv4) which is a widely used
protocol in data communication over different kinds of networks as a delivery of
packets.
• IPv4 protocol has the responsibility of identifying hosts based upon their logical
addresses and to route data among them over the underlying network.
• IPv4 is a connectionless protocol for a packet switched network. Datagrams sent by the
same source to the same destination could arrive out of order. IPv4 relies on the
higher layer for reliability.
3.5.2 IPv4 Datagram Format

• In IPv4, packets are called datagram, which is a basic transfer unit associated with a
packet-switched network.
• The IP datagram format is shown in Fig. 3.32.
Fig. 3.32: IP Datagram
• A datagram is a variable-length packet consisting of two parts namely, header and

data.
• The header is 20 to 60 bytes in length and contains information essential to routing
and delivery.
Header Format:
Fig. 3.33: Header Format
3.33
• A 20-byte header contains almost 13 multipurpose fields, which hold specific related
object information such as application, data type and source/destination addresses.
• IP header format contains following fields:
1. Version (VER): This 4-bits field defines the current version of IP protocol.
Currently the IP version is 4 (IPv4). This field tells the IPv4 software running in the
processing machine that the datagram has format of version 4 and all fields must
be treated as version 4. However, IP version 6 (IPv6) may replace version 4 in the
future.
2. Header Length (HLEN): This 4-bits field defines the total length of the header. The
length of the header is variable (between 20 to 60 bytes). When there are no
options, the header length is 20 bytes, and value is 5(5 × 4 =20). When the option
field is at its maximum size, the value of this field is 15(15 × 4 = 60).
3. Service Type: This 8-bits field previously known as service type, is now called
differentiated services. Both implementations are shown in Fig. 3.34.
D: Minimize delay R: Maximize reliability

T: Maximize throughout C: Minimize cost
D T R C
Precedence TOS bits Codepoint
(a) Service Type (b) Differentiated Services

Fig. 3.34
(i) Service type: First 3 bits are called precedence bits, the next 4 bits are Type Of
Services (TOS) bits and the last bit is not used.
(a) Precedence: This 3-bits subfield ranging from 0(000 in binary) to 7(111 in
binary). This field defines priority of the datagram in issues such as congestion.
If a router is congested, it discards some datagrams. Datagrams with the lowest
priority are discarded first.
(b) TOS Bits: This 4-bits subfield defines types of services. We can have 5 different
types of services as listed in the following table.
Table 3.3: Types of Services
TOS Bits Description

0000 Normal (default)
0001 Minimize cost
0010 Maximize reliability
0100 Maximize throughput
1000 Minimize delay
3.34
(ii) Differentiated Services: The first 6 bits made the code-point subfield, and the last
2 bits are not used.
4. Total length: This 16-bits field defines the total length of the datagram including
the header. IPv4 datagram maximum size is 65,535, of which 20 to 60 bytes are the
header and the rest is data from the upper layer.
Length of data = Total length – Header length
5. Identification: This 16-bits field contains a specific number for primary data
identification. Uniquely identifies the datagram. Usually incremented by 1 each
time a datagram is sent. All fragments of a datagram contain the same
identification value. This allows the destination host to determine which fragment
belongs to which datagram. This field is used in fragmentation.
6. Flags (3-bits): As required by the network resources, if the IP packet is too large to
handle, these ‘flags’ tell if they can be fragmented or not. This router fragment
activity is controlled by following three flags:
Sr. No. Flag Description
1. 0 Reserved, must be zero.
DF (Do not 0 means allow fragmentation;
2.
Fragment) 1 means do not allow fragmentation.
0 means that this is the last fragment of the
MF (More
3. datagram;
Fragments)
1 means that additional fragments will follow.
7. Fragmentation Offset: This 13-bits offset tells the exact position of the fragment in
the original IP packet. This field is used in fragmentation.
8. Time To Live (TTL) (8 bit): Every datagram has a limited lifetime on the Internet.
This field is used to control the maximum number of hops (routers) visited by the
datagram. To avoid looping in the network, every packet is sent with some TTL
value set, which tells the network how many routers (hops) this packet can cross.
At each hop, its value is decremented by one and when the value reaches zero, the
packet is discarded.
9. Protocol: This 8-bits field defines the higher level protocol that uses the services of
IPv4. IPv4 protocol carries data from different other protocols (TCP, UDP, ICMP,
etc.), the value of this field helps the receiving network layer know to which
protocol the data belong.
10. Header Checksum: This 18-bits field is used for error detection.
11. Source address: This 32-bits field defines the source address of a datagram.
12. Destination address: This 32-bits field defines the destination address of a
datagram.
13. Options: This optional field may contain values for options such as Security,
Record Route, Time Stamp, etc.
3.35
3.5.3 Fragmentation (Oct. 17, 18)

• To reach up to destination, datagram may travel through different networks. Every
router decapsulates the IPv4 datagram from the frame it receives, processes it, and
then encapsulates it in another frame.
• The format and size of a frame depends upon the type of a network. Two networks
may have different frame formats and different sizes.
• For example, if a router connects a LAN to a WAN, it receives a frame in the LAN
format and sends a frame in the WAN format.
• Each data link layer protocol has its own frame format in most protocols. Maximum
Transfer Unit (MTU) defines the maximum size of the data field. The value of the MTU
depends on the physical network protocol.
Fig. 3.35: Maximum Transfer Unit (MTU)
Table 3.4: MTU for some networks

Protocol MTU
Hyper-channel 65,535
Token ring (16 mbps) 17,914
Token ring (4 mbps) 4,464
FDDI 4,352
Ethernet 1,500
X.25 576
PPP 296
• To make the IPv4 protocol independent of the physical network, length of IPv4
datagram equal to 65,535 bytes. This makes a transmission more efficient if we use a
protocol with an MTU of this size.
• However, for other physical networks, we must divide the datagram, so that it can
pass through these networks. This is called fragmentation.
• When a datagram is fragmented, every fragment has its own header. Most of the fields
are repeated and some are changed in fragments.
• A fragmented datagram can be fragmented if required. A datagram can be fragmented
several times before it reaches the final destination.
• In IPv4, datagram can be fragmented by the source or routers in the path. But
reassembly of datagram is done only by the destination host.
3.36
• Because every fragmented datagram may be routed independently by different routes

and we can never control or guarantee which route a fragmented datagram may take.
• All these fragments arrive at the destination host. So the reassembly is done at the
final destination.
• The host or router that fragments a datagram must change the values of three fields
i.e., Flags, fragmentation offset and total length. Other fields are copied as it is.
Fields Related to Fragmentation:
• The fields related to fragmentation are identification, flags and fragmentation offset.
These fields are described below:
1. Identification:
• This 16-bits field identifies a datagram originating from the source host. Identification
and source address uniquely defines a datagram as it leaves the source host. For
uniqueness, IPv4 protocol uses a counter.
• When datagram is sent, IPv4 copies the current value of the counter to the
identification field and increments the counter by 1.
• When a datagram is fragmented, this value is copied to all fragments so that all
fragments have the same identification number. This identification number helps the
destination host at the time of reassembly.
2. Flags:
• From the 3-bits, the first bit is reserved.
• The second bit is called the do not fragment bit. If its value is 1, the machine must not
fragment the datagram. If it cannot pass the datagram, it must discard it. If the value is
0, the datagram can be fragmented if necessary.
• The third bit is called the more fragment bit. If its value is 1, means the datagram is
not the last fragment, more fragments are coming after this. If its value is 0, it means
this is the only or last fragment.
D : Do not fragments
D M
M : More fragments
Fig. 3.36: Flags used in Fragmentation
3. Fragmentation Offset:
• This 13-bits field shows the position of the fragment with respect to the whole
datagram.
• Fig. 3.37 shows a datagram with a data size of 4000 bytes fragmented into three
fragments.
• The bytes in the original datagram are 0 to 3999. The first fragment carries 0 to 1399
0
bytes. The offset for this datagram is 8 = 0. The second fragment carries 1400 to 2799,
1400
the offset is 8 = 8 = 175. The third one carries 2800 to 3999 bytes. The offset value is
2800
8 = 350.
3.37
Offset = 0000/8 = 0
0000 1399
Offset = 0000/8 = 0
Offset = 1400/8 = 175
1400 2799
Byte 0000 Byte 3999
Offset = 2800/8 = 350
2800 3999
Fig. 3.37: Example of Fragmentation
• Fig. 3.38 shows an expanded view of the fragments in Fig. 3.37.
1420
14,567 1 000
820
Bytes 0000-1399 14,567 1 175
Fragment 1
4020
14,567 0 000 1420 Bytes 1400-2199
14,567 1 175
Fragment 2.1
620
Bytes 0000-3999 14,567 1 275
Bytes 1400-2799
Original datagram
Fragment 2
1220 Bytes 2200-2799

14,567 0 350
Fragment 2.2
Bytes 2800-3999
Fragment 3
Fig. 3.38: Detailed Example of Fragmentation
3.38
3.5.4 Options
• The header of the IP datagram is made of two parts namely, a fixed part and a variable
part.
• The fixed part is 20 bytes long, and the variable part comprises the options, which can
be a maximum of 40 bytes.
• Options, as the name implies, are not required for a datagram. They can be used for
network testing and debugging.
• Fig. 3.39 shows the format of an option.
Fig. 3.39: Option Format
• The various field in option format are explained below:

1. Type: The type field is 8-bits long and contains three subfields namely, copy, class,
and number.
2. Copy: This 1-bits subfield controls the presence of the option in fragmentation.
When its value is 0, it means that the option must be copied only to the first
fragment. If its value is 1, it means the option must be copied to all fragments.
3. Class: This 2-bits subfield defines the general purpose of the option. When its
value is 00, it means that the option is used for datagram control. When its value is
10, it means that the option is used for debugging and management. The other two
possible values (01 and 11) have not yet been defined.
4. Number: This 5-bits subfield defines the type of option. Although 5 bits can define
up to 32 different types, currently only 6 types are in use.
5. Length: The length field defines the total length of the option including the type
field and the length field itself. This field is not present in all of the option types.
6. Value: The value field contains the data that specific options require. Like the
length field, this field is also not present in all option types.
3.39
Options Types:
• Fig. 3.40 shows types of options.
• Options are not required for a datagram, they are used for network testing and
debugging.
No operation
Single-byte
End of option
Options
Record route
Strict source route

Multiple-byte
Loose source route
Timestamp
Fig. 3.40: Options in IPv4

• Various options in IPv4 are explained below:
1. No Operation Option: A no-operation option is a 1-byte option used as a filler
between options.
2. End of Option: An end-of-option option is a 1-byte option used for padding at the
end of the option field. It can only be used as the last option.
3. Record Route Option: A record route option is used to record the Internet routers
that handle the datagram. It can list up to nine router addresses.
4. Strict Source Route Option: A strict source route option is used by the source to
predetermine a route for the datagram as it travels through the Internet. The
sender can choose a route with a specific type of service (e.g. minimize delay,
maximum throughput). If a datagram specifies a strict source route, all the routers
defined in the option must be visited by the datagram. If an address of a router is
not mentioned in the route list, it must not be visited. If datagram visits a route
that is not on the list, datagram is discarded. If datagram arrives at the destination
and some of the entries were not visited, it will be also discarded.
5. Loose Source Route Option: A loose source route option is similar to strict source
route. Only one difference is, each router in the list must be visited, but the
datagram can visit other routers also.
6. Timestamp Option: A timestamp option is used to record the time of datagram
processing by a router. This can help users and managers to track the behavior of
the routers on the Internet. We can estimate time taken for a datagram to go from
one router to another.
3.40
3.6 MOBILE IP (Oct. 14)

• In the last decade, mobile communication has received a lot of attention. The interest
in mobile communication on the Internet means that the IP protocol, originally
designed for stationary devices, must be enhanced to allow the use of mobile
computers, computers that move from one network to another.
• Mobile IP (MIP) is an Internet Engineering Task Force (IETF) standard
communications protocol that is designed to allow mobile device users to move from
one network to another while maintaining a permanent IP address.
• The goals of a Mobile IP include supporting end-system mobility while maintaining
scalability, efficiency, and compatibility in all respects with existing applications and
Internet protocols.
• Mobile IP communication protocol refers to the forwarding of Internet traffic with a
fixed IP address even outside the home network. It allows users having wireless or
mobile devices to use the Internet remotely.
• Mobile IP for IPv4 is described in IETF RFC 5944, and extensions are defined in IETF
RFC 4721.
• Mobile IPv6, the IP mobility implementation for the next generation of the Internet
Protocol, IPv6 is described in RFC 6275.
3.6.1 Architecture of Mobile IP

• The objective of IP mobility is to maintain the TCP connection between a mobile host
and a static host while reducing the effects of location changes while the mobile host is
moving around, without having to change the underlying TCP/IP.
• The Mobile IP allows for location-independent routing of IP datagrams on the Internet.
Each mobile node is identified by its home address disregarding its current location in
the Internet.
• While away from its home network, a mobile node is associated with a care-of address
which identifies its current location and its home address is associated with the local
endpoint of a tunnel to its home agent.
• Mobile IP specifies how a mobile node registers with its home agent and how the
home agent routes datagrams to the mobile node through the tunnel.
• A mobile node has two addresses, a permanent home address and a Care-of Address
(CoA), which is associated with the network the mobile node is visiting.
• The home network of a mobile device is the network within which the device receives
its identifying IP address (home address). The home address of a mobile device is the
IP address assigned to the device within its home network.
• The network node that is responsible for forwarding and managing this transparency
is known as the home agent.
3.41
• A Home Agent (HA) stores information about mobile nodes whose permanent home
address is in the home agent's network.
• The HA acts as a router on a Mobile Host's (MH) home network which tunnels
datagrams for delivery to the MH when it is away from home, maintains a Location
Directory (LD) for the MH.
• A Foreign Agent (FA) stores information about mobile nodes visiting its network.
Foreign agents also advertise care of addresses, which are used by Mobile IP.
• If there is no foreign agent in the host network, the mobile device has to take care of
getting an address and advertising that address by its own means.
• A Mobile Node (MN) is responsible for discovering whether it is connected to its home
network or has moved to a foreign network. HA’s and FA’s broadcast their presence
on each network to which they are attached.
• A foreign network is the network in which a mobile node is operating when away
from its home network.
• Whenever the mobile node moves, it registers its new care of address with its home
agent. The home agent forwards the packet to the foreign network using the care of
address.
• The delivery requires that the packet header is modified so that the care of address
becomes the destination IP address.
• This new header (See Fig. 3.42) encapsulates the original packet, causing the mobile
node's home address to have no impact on the encapsulated packet's routing. This
phenomenon is called tunneling.
• Fig. 3.41 shows architecture of Mobile IP.
Fig. 3.41: Mobile IP Architecture
3.42
• Let us take an example of IP datagrams being exchanged over a TCP connection

between the mobile node (X) and another host (server M in Fig. 3.41), the following
steps occur:
Step 1: Server M wants to transmit an IP datagram to node X. The home address of X is
advertised and known to M. M does not know whether X is in the home
network or somewhere else. Therefore, M sends the packet to X with X's home
address as the destination IP address in the IP header. The IP datagram is
routed to X's home network.
Step 2: At the X's home network, the incoming IP datagram is intercepted by the home
agent. The home agent discovers that X is in a foreign network. A care of
address has been allocated to X by this foreign network and available with the
home agent. The home agent encapsulates the entire datagram inside a new
IP datagram, with X's care of address in the IP header. This new datagram
with the care of address as the destination address is retransmitted by the
home agent.
Fig. 3.42: The IP Headers in Mobile IP (IP Encapsulation)
3.43
Step 3: At the foreign network, the incoming IP datagram is intercepted by the foreign
agent. The foreign agent is the counterpart of the home agent in the foreign
network. The foreign agent strips off the outer IP header, and delivers the
original datagram to X.
Step 4: A intends to respond to this message and sends traffic to M. In this
example, M is not mobile; therefore M has a fixed IP address. For routing X’s
IP datagram to M, each datagram is sent to some router in the foreign
network. Typically, this router is the foreign agent. X uses M's IP static address
as the destination address in the IP header.
Step 5: The IP datagram from X to M travels directly across the network, using M’s IP
address as the destination address.
• To support the operations illustrated in the previous example, mobile IP needs to
support following three basic capabilities/functions:
1. Discovery: A mobile node uses a discovery procedure to identify prospective
home agents and foreign agents.
2. Registration: A mobile node uses a registration procedure to inform its home
agent of its care-of address.
3. Tunneling: Tunneling procedure is used to forward IP datagrams from a home
address to a care-of address.
3.6.2 Addressing
• The main problem that must be solved in providing mobile communication using the
IP protocol is addressing.
Stationary Hosts:
• The original IP addressing was based on the assumption that a host is stationary,
attached to one specific network. A router uses an IP address to route an IP datagram.
• An IP address has two parts namely, a prefix and a suffix. The prefix associates a host
to a network. For example, the IP address 10.3.4.24/8 defines a host attached to the
network 10.0.0.0/8. This implies that a host on the Internet does not have an address
that it can carry with itself from one place to another.
• The address is valid only when the host is attached to the network. If the network
changes, the address is no longer valid.
• Routers use this association to route a packet; they use the prefix to deliver the packet
to the network to which the host is attached. This scheme works perfectly with
stationary hosts.
1. Mobile Hosts: When a host moves from one network to another, the IP addressing
structure needs to be modified. The solutions are:
3.44
2. Changing the Address: One simple solution is to let the mobile host change its
address as it goes to the new network. The host can use DHCP to obtain a new
address to associate it with the new network. This approach has several
drawbacks.
(i) First, the configuration files would need to be changed.
(ii) Second, each time the computer moves from one network to another, it must
be rebooted.
(iii) Third, the DNS tables need to be revised so that every other host in the
Internet is aware of the change.
(iv) Fourth, if the host roams from one network to another during a transmission,
the data exchange will be interrupted.
• This is because the ports and IP addresses of the client and the server must remain
constant for the duration of the connection.
Two Addresses:
• The approach that is more feasible is the use of two addresses. The host has its original
address, called the home address, and a temporary address, called the care-of address.
• The home address is permanent; it associates the host to its home network, the
network that is the permanent home of the host.
• The care of address is temporary. When a host moves from one network to another,
the care-of address changes; it is associated with the foreign network, the network to
which the host moves.
Fig. 3.43: Home Address and Care of Address (CoA)
3.6.3 Agents
• Mobile IP enables the transfer of information to and from mobile computers, such as
laptops and wireless communications.
• To make the change of address transparent to the rest of the Internet requires a home
agent and a foreign agent.
3.45
• Fig. 3.44 shows the position of a home agent relative to the home network and a
foreign agent relative to the foreign network.
• Home network is a network to which the mobile node originally belongs to as per its
assigned IP address (home address). Foreign Network is the current network to which
the mobile node is visiting (away from its home network).
• Fig. 3.44 shows Home Agent (HA) and Foreign Agent (FA).
1. Home Agent: The home agent is usually a router attached to the home network of
the mobile host. The HA acts on behalf of the mobile host when a remote host
sends a packet to the mobile host. The home agent receives the packet and sends it
to the foreign agent.
2. Foreign Agent: The foreign agent is usually a router attached to the foreign
network. The FA receives packets sent by the home agent and deliver them to the
mobile host.
Fig. 3.44: Home Agent (HA) and Foreign Agent (FA)
3.6.4 Three Phases

• To communicate with a remote host, a mobile host goes through three phases namely,
agent discovery, registration and data transfer.
• Fig. 3.45 shows the three phases, agent discovery (involves the mobile host, the foreign
agent and the home agent) registration (also involves the mobile host and the two
agents) and data transfer (remote host is also involved).
• The Mobile IP process has three main phases as explained below:
Phase I: Agent Discovery:
• This is the first phase of the mobile IP process, where mobile nodes discover its foreign
agents and home agents.
• The agent discovery phase, consists of following two sub phases:
1. A mobile host must discover (learn the address of) a home agent before it leaves its
home network.
3.46
2. A mobile host must also discover a foreign agent after it has moved to a foreign
network. This discovery consists of learning the care of address as well as the
foreign agent’s address.
• The discovery involves two types of messages namely, advertisement and solicitation.
Fig. 3.45: Remote Host and Mobile Host Communication
Agent Advertisement:
• When a router advertises its presence on a network using an ICMP router
advertisement, it can append an agent advertisement to the packet if it acts as an
agent.
• Fig. 3.46 shows how an agent advertisement is piggybacking to the router
advertisement packet.
• Mobile IP does not use a new packet type for agent advertisement, it uses the router
advertisement packet of ICMP (Internet Control Message Protocol) and appends an
agent advertisement message.
• The field descriptions of agent advertisement in Fig. 3.46 are as follows:
1. Type (8-bit) field is set to l6.
3.47
2. Length (8-bit) field defines the total length of the extension message (not the
length of the ICMP advertisement message).
3. Sequence Number (16-bit) field holds the message number. The recipient can use
the sequence number to determine if a message is lost.
4. Lifetime field defines the number of seconds that the agent will accept requests. If
the value is a string of 1s, the lifetime is infinite.
5. Code field is an 8-bit flag in which each bit is set (1) or unset (0). The meanings of
the bits are shown in following table:
Bit Meaning
0 Registration required. No collocated care of addresses.
1 Agent is busy and does not accept registration at this moment.
2 Agent acts as a Home Agent (HA).

3 Agent acts as a Foreign Agent (FA).
4 Agent uses minimal encapsulation.
5 Agent uses Generic Routing Encapsulation (GRE).

6 Agent supports header compression.
7 Unused (0).
6. Care-of Addresses field contains a list of addresses available for use as care of
addresses. The mobile host can choose one of these addresses. The selection of this
care-of address is announced in the registration request.
ICMP
Advertisement message
Type Length Sequence Number
Lifetime Code Reserved
Care of Address (CoA)

(Foreign Agent (FA) only)
Fig. 3.46: Fields of Agent Advertisement
Agent Solicitation:
• When a mobile host has moved to a new network and has not received agent
advertisements, it can initiate an agent solicitation by using the ICMP solicitation
message to inform an agent that it needs assistance.
3.48
• Mobile IP does not use a new packet type for agent solicitation, it uses the router
solicitation packet of ICMP.
Phase II: Registration:
• This is the second phase in the Mobile IP process, where a mobile node registers its
current location with the foreign agent and the home agent.
• After a mobile host has moved to a foreign network and discovered the foreign agent,
it must register.
• There are following four aspects of registration:
1. The mobile host must register itself with the foreign agent.
2. The mobile host must register itself with its home agent. This is normally done by
the foreign agent on behalf of the mobile host.
3. The mobile host must renew registration if it has expired.
4. The mobile host must cancel its registration (deregistration) when it returns home.
• To register with the foreign agent and the home agent the mobile host uses a
registration request and a registration reply as shown in Figs. 3.47 and 3.48.
Registration Request:
• A registration request is sent from the mobile host to the foreign agent to register its
cartel address and also to announce its home address and home agent address.
• The foreign agent, after receiving and registering the request, relays the message to
the home agent.
• Note that the home agent now knows the address of the foreign agent because the IP
packet that is used for relaying has the IP address of the foreign agent as the source
address.
• Fig. 3.47 shows the format of the registration request.
Type Flag Lifetime
Home address
Home agent address
Care of address
Identification
Extensions
Fig. 3.47: Registration Request Format
3.49
• The field descriptions of registration request in Fig. 3.47 are as follows:

1. Type (8-bit) field defines the type of the message. For a request message the value
of this field is 1.
2. Flag (8-bit) field defines forwarding information. The value of each bit have a
meaning given in following table:
Bit Meaning
0 Mobile host requests that home agent retain its prior care of address.
1 Mobile host requests that home agent tunnel any broadcast message.
2 Mobile host is using collocated care of addresses.
3 Mobile host requests that home agent use minimal encapsulation.
4 Mobile host requests Generic Routing Encapsulation (GRE).
5 Mobile host requests header compression.
6-7 Reserved bits.
3. Lifetime field defines the number of seconds the registration is valid. If the field is
a string of 0s, the request message is asking for deregistration. If the field is a
string of 1s, the lifetime is infinite.
4. Home Address field contains the permanent (first) address of the mobile host.
5. Home Agent Address contains the address of the home agent.
6. Care of Address field is temporary (second) address of the mobile host.
7. Identification field contains a 64-bit number that is inserted into the request by
the mobile host and repeated in the reply message. It matches a request with a
reply.
8. Extensions field used for authentication. Variable length extensions allow a home
agent to authenticate the mobile agent.
Registration Reply:
• A registration reply is sent from the home agent to the foreign agent and then relayed
to the mobile host.
• The reply confirms or denies the registration request. Fig. 3.48 shows the format of the
registration reply.
Type Code Lifetime
Home address
Home agent address
Identification
Extensions
Fig. 3.48: Registration Reply Format
3.50
• The fields are similar to those of the registration request with the exceptions like the
value of the type field is 3, the code field replaces the flag field and shows the result of
the registration request (acceptance or denial). and the care-of address field is not
needed.
• Registration messages are encapsulated in a UDP user datagram. An agent uses the
well-known part 434; a mobile host uses an ephemeral port.
Phase III: Data Transfer:
• After agent discovery and registration, a mobile host can communicate with a remote
host.
• Fig. 3.49 shows the communication idea.
Fig. 3.49: Data Transfer Phase in Mobile IP

From Remote Host to Home Agent:
• When a remote host wants to send a packet to the mobile host, it uses its address as
the source address and the home address of the mobile host as the destination
address.
• In other words, the remote host sends a packet as though the mobile host is at its home
network.
• The packet, however, is intercepted by the home agent, which pretends it is the mobile
host. This is done using the proxy ARP technique.
• Path 1 of Fig. 3.49 shows this step.
3.51
From Home Agent to Foreign Agent:

• After receiving the packet, the home agent sends the packet to the foreign agent using
the tunneling.
• The home agent encapsulates the whole IP packet inside another IP packet using its
address as the source and the foreign agent’s address as the destination.
From Foreign Agent to Mobile Host:
• When the foreign agent receives the packet, it removes the original packet.
• However, since the destination address is the home address of the mobile host, the
foreign agent consults a registry table to find the address of the mobile host,
(otherwise, the packet would just be sent back to the home network).
• The packet is then sent to the care of address. Path 3 of Fig. 3.49 shows this step.
From Mobile Host to Remote Host:
• When a mobile host wants to send a packet to a remote host (for example, a response
to the packet it has received), it sends as it does normally.
• The mobile host prepares a packet with its home address as the source, and the
address of the remote host as the destination.
• Although the packet comes from the foreign network, it has the home address of the
mobile host.
3.7 NEXT GENERATION IP

• The network layer protocol in the TCP/IP protocol suite is currently IPv4
(Internetworking Protocol version 4).
• IPv4 has some deficiencies (like inefficient address space (32-bits), inefficient for audio
and video data, no encryption and authentication security facility etc.) that make it
unsuitable for the fast-growing Internet.
• To overcome these deficiencies IPv6 (Internetworking Protocol version 6) was
proposed and is now a standard. IPv6 is the specific protocol chosen by the IETF
(Internet Engineering Task Force) as the Internet's Next Generation IP.
• IPv6 is also called IPng (Internetworking Protocol, Next Generation). It has a 128-bit
address space. IPv6 also includes addressing and security features.
• The next generation IP had many advantages on the previously existing version of
Internet protocol (IPv4). These are listed below:
1. Larger Address Space: An IPv6 address is 128 bits long (compared to IPv4, IPv6
address is very long because IPv4 address was only 32 bits).
3.52
2. Better Header Format: IPv6 uses a new header format in which options are
separated from the base header and inserted when needed, between base header
and upper layer data. This simplifies and speeds up the routing process because
most of the options do not need to be checked by routers.
3. New Options: IPv6 has new options to allow additional functionalities.
4. Allowance for Extension: IPv6 is designed to allow the extension of the protocol if
required by new technologies or applications.
5. Support for Resource Allocation: In IPv6, the type of service field has been
removed, but a mechanism (called flow label) has been added to enable the source
to request special handling of packets. This mechanism can be used to support
traffic such as real time audio and video.
6. Support for More Security: The encryption and authentication options in IPv6
provide confidentiality and integrity of the packet.
7. Plug and Play: IPv6 includes plug and play in the standard specification. It
therefore must be easier for novice users to connect their machines to network, it
will be done automatically.
8. Clearer Specification: IPv6 follows good practices of IPv4, and rejects its minor
problems.
3.7.1 IPv6 Address Representation

• An IPv6 address consists of 16 bytes (octets); it is 128 bits long, as shown in Fig. 3.50.
128 bits = 16 bytes
111 0 11 0 11 0 11111 1111111111
ABFC 1234 13FB 11 3 B 1C3B FBAB 12FC FFFF
Fig. 3.50: IPv6 Address
Hexadecimal Colon Notation:

• To make addresses more readable, IPv6 specifies hexadecimal colon notation. In this
notation, 128 bits are divided into eight sections, each of 2 bytes in length.
• Two bytes in hexadecimal notation require four hexadecimal digits. Therefore, the
address consists of 32 hexadecimal digits, with every four digits separated by a
colon (:).
3.53
Abbreviation :
• Although IP addresses in hexadecimal format are very long, many of the digits are
zeros, in this case we can abbreviate the address.
• The leading zeros of the section (four digits between two colons) can be omitted. Only
leading zeros can be dropped, not the trailing zeros, (Refer Fig. 3.51).
Unabbreviated
FDEC BA98 0007 3210 000F 0000 0002 FFFF
FDEC BA98 7 3210 F 0 2 FFFF
Abbreviated
Fig. 3.51: Abbreviated Address
• Using this form of abbreviation, 0007 can be written as 7, 000F as F, and 0000 as 0.
Note that 3210 cannot be abbreviated.
• Further abbreviation is possible, if there are consecutive sections consisting of zeros
only.
• We can remove the zeros altogether and replace them with double semicolons. Refer
to the following Fig. 3.52. Note that this type of abbreviation is allowed only once per
address. If there are two runs of zero sections, only one of them can be abbreviated.
Abbreviated
ABFC 0 0 0 0 BBFF 0 FFFF
ABFC BBFF 0 FFFF
More abbreviated
Fig. 3.52: Abbreviated Address with Consecutive Zeros
• Re-expansion of the abbreviated address is very simple: align the unabbreviated

portions and insert zeros to get the original expanded address.
3.7.2 IPv6 Address Space

• The address space of IPv6 is much larger as compared to IPv4.
3.54
128
• The address space of IPv6 contains 2 addresses as shown below and this address
96
space is 2 times of the IPv4 address,
340,282,366,920,938,463,374,607,431,768,211,456
3.7.3 IPv6 Address Types

• The addressing architecture of IPv6 is defined in RFC 4291 and allows three different
types of transmission namely, unicast, anycast and multicast.
Fig. 3.53: Types of IPv6 Addressing

1. Unicast Address:
• A unicast address defines a single computer or router.
• The packet sent to the unicast address must be delivered to that specific/intended
computer.
Fig. 3.54: Unicast Addressing
• IPv6 defines two types of unicast addresses namely, geographically based and
provider based.
• A provider based address is generally used. The fields for provider based address are
type identifier, Registry identifier, provider identifier, subscriber identifier, subnet
identifier, and node identifier.
2. Anycast Address:
• An anycast address is assigned to multiple interfaces (usually on multiple nodes). An
anycast address defines a group of computers that all share a single address.
• A packet sent to an anycast address must be delivered to exactly one of the members
of the group – the closest or the most easily accessible.
3.55
Fig. 3.55: Anycast Addressing

3. Multicast Address:
• It defines a group of computers. In multicast address each member of the group
receives a copy.
• The packet sent to a multicast address must be delivered to each member of the group.
Fig. 3.56: Multicast Addressing
3.7.4 IPv6 Protocol

• Internet Protocol version 6 (IPv6) is a network layer protocol that enables data
communications over a packet switched network.
• IPv6 is also called IPng (Internetworking Protocol next generation).
• Like IPv4, IPv6 is a connectionless, unreliable datagram protocol that is primarily
responsible for addressing and routing packets between hosts.
• IPv6 is the most recent version of the Internet Protocol (IP), the communications
protocol that provides an identification and location system for computers on
networks and routes traffic across the Internet.
3.56
Features of IPv6:
1. Expanded Addressing Capabilities: IPv6 increases the IP address size from 32
bits to 128 bits, to support more levels in the addressing hierarchy, a much greater
number of addressable nodes, and simpler auto-configuration of addresses.
2. Support for Resource Allocation: In IPv6, the type-of-service field has been
removed, but two new fields, traffic class and flow label have been added to
enable the source to request special handling of the packet. This mechanism can
be used to support traffic such as real-time audio and video.
3. Support for More Security: The encryption and authentication options in IPv6
provide confidentiality and integrity of the packet. IPv6 basic specification
includes security in the form of packet encryption and source authentication
4. Header Format Simplification: Some IPv4 header fields have been dropped or
made optional, in IPv6 to reduce the common-case processing cost of packet han-
dling and to limit the bandwidth cost of the IPv6 header.
5. New Options: IPv6 has new options to allow for additional functionalities.
6. Auto-configuration: lPv6 basic specification includes address auto-configuration.
So, even a novice user can have conned his/her machine to the Internet.
7. Allowance for Extension: IPv6 is designed to allow the extension of the protocol if
required by new technologies or applications.
8. Scalability: IPv6 uses 128-bits address space and this address length is 4 times
longer than IPv4 (32 bits). Thus many more IP addresses can be added to the
Internet.
9. Mobility: IPv6 was designed keeping mobility in mind. This feature enables hosts,
(such as mobile phones) to roam around in different geographical areas and
remain connected with the same IP address. The mobility feature of IPv6 takes
advantage of auto IP configuration and Extension headers.
10. End to End Connectivity: Every system now has a unique IP address and can
traverse through the Internet without using NAT or other translating components.
After IPv6 is fully implemented, every host can directly reach other hosts on the
Internet, with some limitations involved like Firewall, organization policies, etc.
3.7.5 IPv6 Packet Format

• Fig. 3.57 shows the datagram/packet format of IPv6 protocol. Each packet in IPv6
consists of a mandatory base header followed by the payload.
• The payload includes two parts namely, optional extension headers and data from an
upper layer.
• The base header consumes 40 bytes, inversely the extension headers and data from
the top layer usually hold up to 65,535 bytes of information.
3.57
40 bytes Up to 65,535 bytes
Base header Payload
Extension headers
Data packet from upper layer
(optional)
Fig. 3.57: IPv6 Datagram/Packet Format
Base Header:
• Fig. 3.58 shows the base header with its eight fields of IPv6 datagram.
32 bits
Version Traffic Class Flow Label

0-3 4-11 12-31
Payload Length Next Header Hop Limit
32-47 48-55 56-63
Source Address
64-191
Destination Address
192-288
Fig. 3.58: Base Header Format of IPv6

• The various fields in IPv6 base header format are explained below:
1. Version (4 bits) field specifies the version of Internet Protocol number. For IPv6 it
is 6 i.e. 0110.
2. Traffic Class (4 bits) field defines the priority of the packet with respect to traffic
congestion.
3. Flow Label (20 bits) field that is designed to provide special handling for a
particular flow of data. The purpose of flow label field is to indicate that the packet
belongs to a specific sequence of packets between source to destination and can be
used to prioritized delivery of packets for services like voice.
4. Payload Length (16 bits) is a field defining the total length of IP datagram
including the base header.
5. Next Header (8 bits) field identifies the type of header immediately following the
IPv6 header.
6. Hop Limit (8 bits) field serves the same purpose as the TTL (Time To Live) field in
IPv4. The Hop Limit field shows the maximum number of routers the IPv6 packet
can travel.
3.58
7. Source Address (128 bits) field identifies the original source of the datagram.
8. Destination Address (128 bits) field identifies the destination of the datagram.
3.7.6 Extension Header

• In IPv6, the fixed header contains only that much information which is necessary,
avoiding that information which is either not required or is rarely used.
• All such information is put between the fixed header and the Upper layer header in
the form of extension headers. Each extension header is identified by a distinct value.
• Extension headers allow additional functionality to be implemented in an IPv6 packet.
These fields are used only for specific purposes. This permits the IPv6 packet to
remain small and streamlined and possess only the fields that are required for its
particular purpose.
• Fig. 3.59 shows the extension header format.
Fig. 3.59: Extension Header Format of IPv6
• When Extension Headers are used, IPv6 Fixed Header’s Next Header field points to the
first Extension Header. If there is one more Extension Header, then the first Extension
Header’s ‘Next-Header’ field points to the second one, and so on.
• The last Extension Header’s ‘Next-Header’ field points to the Upper Layer Header.
Thus, all the headers point to the next one in a linked list manner.
• If the Next Header field contains the value 59, it indicates that there are no headers
after this header, not even Upper Layer Header.
• Each extension header is identified by a specific Next Header value, and the most
common values are given in Table 3.5.
3.59
• IPv6 extension headers are optional but, when used, have specific ordering
requirements that must be followed. This prevents the destination node from scanning
the packet and looking for a particular type of extension header so chat header call k
processed ahead of the others.
Table 3.5: Extension Headers must be supported as per RFC
Sr. Next Header
Extension Header Description
No. Value
1. Hop-by-Hop options header 0 Read by all devises in the
transit network.
2. Routing header 43 Contains methods to support
making routing decisions.
3. Fragment header 44 Contains parameters of
datagram fragmentation.
4. Destination Options header 60 Read by destination devices.
5. Authentication header 51 Information regarding
authenticity.
6. Encapsulating Security 50 Encryption information.
Payload (ESP) header
• The sequence of Extension Headers should be:
IPv6 header
Hop-by-Hop Options header

1
Destination Options header
Routing header
Fragment header
Authentication header
Encapsulating Security Payload header

2
Destination Options header
Upper-layer header
Fig. 3.60
• These headers:
1. should be processed by First and subsequent destinations.
2. should be processed by Final Destination.
3.60
• Extension Headers are arranged one after another in a linked list manner, as depicted
in the Fig. 3.61.
Fig. 3.61
3.7.7 Difference Between IPv4 and IPv6

• Following table differentiate between IPv4 and IPv6:
Sr.
IPv4 IPv6
No.
1. In IPv4 source and destination In IPv6 source and destination addresses
addresses are 32 bits (4 bytes) in length. are 128 bits (16 bytes) in length.
2. IPv4 header is 20 bytes. IPv6 header is 40 bytes.
32 128
3. There are a maximum 2 IP addresses. There are maximum 2 IP addresses.
4. IPv4 addresses are written by dotted IPv6 addresses are written in
decimal notation. hexadecimal colon notation.
For example: 10.15.11.23. For example: FADB:A2B2:A453:1212:
AAB3:ADBD:BBCC:1234
5. Checksum available in IPv4 header. No Checksum field in IPv6 header.
6. In IPv4 manual configuration is IPv6 address can be configured
required or need to use DHCP. automatically.
7. In IPv4 the sending host and the router In IPv6 only sending hosts do the
do the fragmentation. fragmentation.
8. IPv4 is being used as less secure IPv6 has its inbuilt security feature
protocol as its security section is named IPSec (Internet Protocol Security).
dependent on application.
9. IPv4 does not provide packet flow IPv6 packet flow identification is
identification. available within the IPv6 header using
the Flow Label field.
10. IPv4 does not provide encryption and IPv6 provides encryption and
authentication. authentication.
11. IPv4 uses unicast, broadcast and IPv6 uses unicast, multicast and anycast
multicast addresses. addresses.
3.61
3.8 ROUTING (April 18)
• When a network device has multiple paths to reach a destination, it always selects one
path by preferring it over others. This selection process is termed as Routing.
• Routing is done by special network devices called routers or it can be done by means
of software processes.
• Routing is the process of establishing the routes that data packets must follow to reach
the destination.
3.8.1 General Idea of Routing (April 18)
• Routing is a process of selecting the path along which the data can be transferred from
source to the destination. Routing is performed by a special device known as a router.
• Routing is a process which is performed by layer 3 (network layer) devices in order to
deliver the packet by choosing an optimal path from one network to another.
• In internetworking, routing the process of moving a packet of data from source to
destination.
• Routing is an activity that transmits information from a source address to a
destination address through an interconnected network.
• When a router receives an IP packet, the router searches its routing table for the best
matching route based on the destination IP address of the packet and forwards the
packet through the outbound interface or next-hop IP address for the route.
• The routing table contains the routing entries (routes) learned by the router in various
ways. The router can obtain routing entries in static or dynamic mode and maintain
its own routing table.
Role of Router:
• A router selects routes and forwards packets. Upon receiving a packet, a router selects
a proper path, which may have one or multiple hops, to send the packet to the next
router according to the destination address in the packet. The last router is responsible
for sending the packet to the destination host.
• A route is a path along which packets are sent from the source to the destination.
When multiple routes are available to send packets from a router to the destination,
the router can select the optimal route from an IP routing table.
• There are following three types of routing:
1. Static routing is a process in which we have to manually add routes in the routing
table.
2. Dynamic routing makes automatic adjustment of the routes according to the
current state of the route in the routing table.
3.62
3. Default routing is the method where the router is configured to send all packets
towards a single router (next hop). It doesn’t matter to which network the packet
belongs, it is forwarded out to the router which is configured for default routing.
Fig. 3.62: Concept of Router
3.8.2 Routing Algorithms

• The routing algorithms are used for routing the packets. The routing algorithm is
nothing but a software responsible for deciding the optimal path through which
packets can be transmitted.
• A routing algorithm is a set of step-by-step operations used to direct Internet traffic
efficiently.
• When a packet of data leaves its source, there are many different paths it can take to
its destination. The routing algorithm is used to determine/choose the best path from
source to destination. Different routing algorithms use different methods to determine
the best path.
• Routing is the process of establishing the routes that data packets must follow to reach
the destination.
• A routing algorithm specifies how packets choose the path to their destinations. There
are two types of routing algorithms namely, deterministic and adaptive.
• In deterministic routing only one path is determined through source to destination,
while adaptive routing algorithms allow multiple paths.
• The main properties of routing are: (April 18)
1. Correctness: The routing should be done properly and correctly so that the
packets may reach their proper destination.
2. Simplicity: The routing should be done in a simple manner so that the overhead is
as low as possible. With increasing complexity of the routing algorithms the
overhead also increases.
3.63
3. Robustness: Once, a major network becomes operative, it may be expected to run

continuously for years without any failures. The algorithms designed for routing
should be robust enough to handle hardware and software failures and should be
able to cope with changes in the topology and traffic without requiring all jobs in
all hosts to be aborted and the network rebooted every time some router goes
down.
4. Stability: The routing algorithms should be stable under all possible
circumstances.
5. Fairness: Every node connected to the network should get a fair chance of
transmitting their packets. This is generally done on a first come first serve basis.
6. Optimality: The routing algorithms should be optimal in terms of throughput and
minimizing mean packet delays. Here there is a trade-off and one has to choose
depending on his suitability.
• The routing algorithm can be classified into following two types:
1. Static (Non-Adaptive) Routing Algorithms:
• In this type of algorithm, the network topology determines the final path. All the
possible paths which are already calculated are loaded into the routing table.
• Static routing is suitable for small networks. The disadvantage of static routing is,
inability to respond quickly in case of network failure.
2. Dynamic (Adaptive) Routing Algorithms:
• The dynamic routing algorithm can change their routing decision on the basis of some
changes made in the topology.
• Each router can check the network status by communicating with the neighbors. So,
the changes in the topology are reflected to all routers.
• Finally, the router can calculate the suitable path to the final destination. The
disadvantage of this type is its complexity in the router.
3.8.3 Types of Routing Algorithms

• In this section we will study various routing algorithms like distance vector routing
algorithm, link state routing algorithm and path-vector routing algorithm.
3.8.3.1 Distance Vector Routing Algorithm

• Distance vector routing algorithm is the dynamic routing algorithm in computer
networks. Distance vector routing algorithm also known as Bellman-Ford routing
algorithm (also called Ford-Fulkerson algorithm) to find the shortest path between
nodes in a graph given the distance between nodes.
• It was designed for small network topologies. Distance Vector Routing (DVR) method
sees an AS, with all routers and networks, as a graph, a set of nodes and lines (edges)
connecting the nodes.
3.64
• A router can normally be represented by a node and a network by a link connecting

two nodes, although other representations are also possible.
• In the distance vector routing algorithm, the node router constructs a table containing
the distance (total cost of path) to all other nodes and distributes that vector to its
immediate neighbors.
• For distance vector routing, it is assumed that each node knows the cost of the link to
each of its directly connected neighbors.
• A link, which is 'down' (which is not working) is assigned as an infinite cost. Every
node sends a message to its directly connected neighbors.
• For example: A sends its information to B and F.
• After communicating to each directly connected node the shortest path can be easy to
compute (See Fig. 3.63).
Fig. 3.63: Distance Vector Routing
• The shortest path can be computed as:
Information Cost to Reach Node

at Node A B C D E F G
A 0 1 2 3 2 1 2
B 1 0 1 2 1 2 3
C 2 1 0 2 1 3 3
D 3 2 2 0 1 2 2
E 2 1 1 1 0 3 2
F 1 2 3 2 3 0 1
G 2 3 3 1 2 1 0
• Distance vector routing protocols are like road signs because routers must make
preferred path decisions based on a distance or metric to a network.
• Just as travelers trust a road sign to accurately state the distance to the next town, a
distance vector router trusts that another router is advertising the true distance to the
destination network.
3.65
• In distance vector routing, the least-cost route between any two nodes is the route with
minimum distance (mi). In this protocol, as the name implies, each node maintains a
vector (table) of minimum distances to every node.
• The table at each node also guides the packets to the desired node by showing the next
stop in the route (next-hop routing).
Fig. 3.64: Distance Vector Routing Tables
• The table for node A in Fig. 3.64 shows how we can reach any node from this node. For
example, our least cost to reach node E is 6. The route passes through C.
• Initialization: The tables in Fig. 3.64 are stable; each node knows how to reach any
other node and the cost. At the beginning, however, this is not the case. Each node can
know only the distance between itself and its immediate neighbors, those directly
connected to it. So for the moment, we assume that each node can send a message to
the immediate neighbors and find the distance between itself and these neighbors.
The distance for any entry that is not a neighbor is marked as infinite (unreachable).
• Sharing: The whole idea of distance vector routing is the sharing of information
between neighbors. Although node A does not know about node E, node C does. So if
node C shares its routing table with A, node A can also know how to reach node E. On
the other hand, node C does not know how to reach node D, but node A does. If node A
shares its routing table with node C, node C also knows how to reach node D. In other
words, nodes A and C, as immediate neighbors, can improve their routing tables if
they help each other.
3.66
Fig. 3.65: Initialization of Tables in Distance Vector Routing (DVR)

• Updating: When a node receives a two-column table from a neighbor, it needs to
update its routing table. Updating takes following three steps:
Step 1: The receiving node needs to add the cost between itself and the sending node
to each value in the second column. The logic is clear. If node C claims that its
distance to a destination is x mi, and the distance between A and C is y mi,
then the distance between A and that destination, via C, is x + y mi.
Step 2: The receiving node needs to add the name of the sending node to each row as
the third column if the receiving node uses information from any row. The
sending node is the next node in the route.
Step 3: The receiving node needs to compare each row of its old table with the
corresponding row of the modified version of the received table.
(i) If the next-node entry is different, the receiving node chooses the row with
the smaller cost. If there is a tie, the old one is kept.
(ii) If the next-node entry is the same, the receiving node chooses the new row.
For example, suppose node C has previously advertised a route to node X
with distance 3.
Fig. 3.66: Updating Distance Vector Routing

3.67
Bellman-Ford Algorithm:
• The Bellman–Ford algorithm is an algorithm that computes shortest paths from a
single source vertex to all of the other vertices in a weighted digraph.
• Bellman-Ford algorithm solves single shortest path problem in which edge weight may
be negative but no negative cycle exists.
• The Bellman–Ford algorithm works correctly when some of the edges of the directed
graph G may have negative weight. When there are no cycles of negative weight, then
we can find out the shortest path between source and destination.
• It is slower than Dijkstra's algorithm but more versatile, as it is capable of handling
some of the negative weight edges. This algorithm detects the negative cycle in a graph
and reports their existence.
• Based on the "Principle of Relaxation" in which more accurate values gradually
recovered an approximation to the proper distance by until eventually reaching the
optimum solution.
• Given a weighted directed graph G = (V, E) with source s and weight function w: E → R,
the Bellman-Ford algorithm returns a Boolean value indicating whether or not there is
a negative weight cycle that is attainable from the source.
• If there is such a cycle, the algorithm produces the shortest paths and their weights.
The algorithm returns TRUE if and only if a graph contains no negative - weight cycles
that are reachable from the source.
Recurrence Relation:
distk [u] = [min[distk-1 [u],min[ distk-1 [i]+cost [i,u]]] as i except u.
k → k is the source vertex
u → u is the destination vertex
i → no of edges to be scanned concerning a vertex.
• Bellman-Ford algorithm can be used in many applications in graph theory. Given a
graph and a source vertex src in the graph, find shortest paths from src to all vertices
in the given graph.
• The graph may contain negative weight edges. Time complexity of this algorithm is
O(VE) which is more than Dijkstra’s algorithm O(VlogV) with the use of Fibonacci
heap.
Input: Graph with source vertex src.
Output: Shortest distance to all vertices from src. If there is a negative weight cycle,
then shortest distances are not calculated, negative weight cycle is reported.
Step 1: This step initializes distances from source to all vertices as infinite and
distance to source itself is 0. Array dis[ ] of size v will keep these values.
3.68
Step 2: This step calculates shortest distances. Do following |v|-1 times.

Do following for each edge u-v
If dist[v] > dist[u] + weight of u-v then
dist[v]=dist[u] + weight of u-v
Step 3: This step reports if there is negative weight cycle in graph
Do the following for each edge u-v
If dist[v]>dist[u] + weight of edge u-v then
“Graph contains negative weight cycle”
• The sequence of steps in Bellman-Ford algorithm are given below:
BELLMAN-FORD(G, w, s)
1. INITIALIZE-SINGLE-SOURCE(G, s)
2. for i ← 1 to |V[G]| - 1
3. do for each edge (u, v) E[G]
4. do RELAX(u, v, w)
5. for each edge (u, v) E[G]
6. do if d[v] > d[u] + w(u, v)
7. then return FALSE
8. return TRUE
Examples:
Example 1: Fig. 3.67 shows a map with nodes and lines and the cost of each line is
given over the line. Find the least cost between the nodes.
Fig. 3.67
Solution: List of edges: (a, b), (a, c), (a, d), (b, e), (c, b), (c, e), (d, c), (d, f), (e, g), (f, g).
Initially:
Node A B C D E F G
Distance 0 ∞ ∞ ∞ ∞ ∞ ∞
Distance From
Fig. 3.68
3.69
Iteration 1:
Node A B C D E F G
Distance 0 3 3 5 5 4 7
Distance From 0 C D A B D F
Fig. 3.69
Iteration 2:
Node A B C D E F G
Distance 0 1 3 5 2 4 5
Distance From 0 C D A B D E
Fig. 3.70
Iteration 3:
Node A B C D E F G
Distance 0 1 3 5 0 4 3
Distance From 0 C D A B D E
Fig. 3.71
Fig. 3.72
Solution: Here, first we list all the edges and their weights.
3.70
Vertices
1 2 3 4 5
No. of edges 1 0 6 5 ∞ ∞
traversed 2 0 3 5 5 8
3 0 3 5 2 8
4 0 3 5 2 5
Fig. 3.73
Solution:
Step 1: Consider A as a source index.
No. of Nodes A B C D E
Distance 0 6 7 ∞ ∞
Distance From A A A
Fig. 3.74
Step 2: Consider Vertex B.
Distance 0 6 7 11 2
Distance From A A A B B
Fig. 3.75
3.71
Step 3: Consider Vertex E.
Distance 0 6 7 9 2
Distance From A A A E B
Fig. 3.76
Step 4: Consider Vertex C.
Distance 0 6 7 4 2
Distance From A A A C B
Fig. 3.77
Step 5: Consider Vertex D.
Distance 0 2 7 4 2
Distance From A D A C B
Fig. 3.78
Step 6: Consider Vertex B.
Distance 0 2 7 4 –2
Fig. 3.79
3.72
Step 7: Consider Vertex E.
Distance 0 2 7 4 –2
Fig. 3.80
Result:
Vertex Distance From A
A 0
B 2
C 7
D 4
E –2
3.8.3.2 Link State Routing Algorithm

• Link state routing is a technique in which each router shares the knowledge of its
neighborhood with every other router in the internetwork.
• Link state routing has a different philosophy from that of distance vector routing.
• In link state routing, if each node in the domain has the entire topology of the domain
the list of nodes and links, how they are connected including the type, cost (metric),
and condition of the links (up or down)-the node can use Dijkstra's algorithm to build a
routing table.
• Link state routing protocols are more like a road map because they create a
topological map of the network and each router uses this map to determine the
shortest path to each network.
• Just as you refer to a map to find the route to another town, link-state routers use a
map to determine the preferred path to reach another destination.
• Routers running a link state routing protocol send information about the state of its
links to other routers in the routing domain.
• The state of those links refers to its directly connected networks and includes
information about the type of network and any neighboring routers on those
networks-hence the name link state routing protocol.
3.73
Fig. 3.81: Concept of Link State Routing (LSR)
Fig. 3.82: Link State Knowledge
• The Fig. 3.82 shows a simple domain with five nodes. Each node uses the same
topology to create a routing table, but the routing table for each node is unique
because the calculations are based on different interpretations of the topology.
• This is analogous to a city map. While each person may have the same map, each
needs to take a different route to reach her specific destination
3.74
Building Routing Tables:

• In link state routing, four sets of actions are required to ensure that each node has the
routing table showing the least-cost node to every other node.
1. Creation of the states of the links by each node, called the link state packet or LSP.
2. Dissemination of LSPs to every other router, called flooding, in an efficient and
reliable way.
3. Formation of a shortest path tree for each node.
4. Calculation of a routing table based on the shortest path tree.
Formation of Shortest Path Tree (Dijkstra Algorithm):
• A tree is a graph of nodes and links; one node is called the root. All other nodes can be
reached from the root through only one single route.
• A shortest path tree is a tree in which the path between the root and every other node
is the shortest. What we need for each node is a shortest path tree with that node as
the root.
• The Dijkstra algorithm is used to create a shortest path tree from a given graph. The
algorithm uses the following steps:
o Initialization: Select the node as the root of the tree and add it to the path. Set the
shortest distances for all the root’s neighbors to the cost between the root and
those neighbors. Set the shortest distance of the root to zero.
o Iteration: Repeat the following two steps until all nodes are added to the path:
1. Adding the next node to the path: Search the nodes not in the path. Select the
one with minimum shortest distance and add it to the path.
2. Updating: Update the shortest distance for all remaining nodes using the
shortest distance of the node just moved to the path in Step 2.
o Dj = minimum (Dj, Di + cij) for all remaining nodes.
Dijkstra Algorithm:
Dijkstra ()
{
// Initialization
Path = {s} // s means self
for (i = 1 to N)
{
If (i is a neighbor of s and I ≠ s) Di= csi
if (i is not a neighbor of s) Di=∞
}
Ds = 0
3.75
} // Dijkstra
// Iteration
Repeat
{
// Finding the next node to be added
Path = Path ∪ i if Di is minimum among all remaining nodes
// Update the shortest distance for the rest
for (j = 1 to M) // M number of remaining nodes
{
Dj = minimum (Dj, Dj + cij)
}
} until (all nodes included in the path, M = 0)
Examples:
Example 1: Fig. 3.83 shows the formation of the shortest path tree for the graph of
seven nodes.
In the initialization step, node A selects itself as the root. It then assigns shortest path
distances to each node on the topology. The nodes that are not neighbors of A receive a
shortest path distance value of infinity.
3.76
Fig. 3.83
Solution: In each iteration, the next node with minimum distance is selected and
added to the path. Then all shortest distances are updated with respect to the last node
selected. For example, in the first iteration, node B is selected and added to the path and
the shortest distances are updated with respect to node B (The shortest distances for C and
E are changed, but for the others remain the same). After six iterations, the shortest path
tree is found for node A. Note that in iteration 4, the shortest path to G is found via C, but
in iteration 5, a new shortest route is discovered (via G); the previous path is erased and
the new one is added.
Calculation of Routing Table from Shortest Path Tree (SPT):
Each node uses the shortest path tree found in the previous discussion to construct its
routing table. The routing table shows the cost of reaching each node from the root.
Routing table for node A using the shortest path tree found in above Fig. 3.84.
Destination Cost Next Router
A 0 –
B 2 –
C 7 B
D 3 –
E 6 B
F 8 B
G 9 B
Fig. 3.84: Routing Table for Node A
3.77
Example 2: Consider Fig. 3.85. Find the shortest path.
Fig. 3.85
Solution:
The set sptSet is initially empty and distances assigned to
vertices are {0, INF, INF, INF, INF, INF, INF, INF} where INF
indicates infinite. Now pick the vertex with minimum distance
value. The vertex 0 is picked, including it in sptSet.
So sptSet becomes {0}. After including 0 to sptSet, update distance
values of its adjacent vertices. Adjacent vertices of 0 are 1 and 7.
Fig. 3.86
The distance values of 1 and 7 are updated as 4 and 8. The
vertices included in SPT are shown in black colour, (See Fig.
3.86).
Pick the vertex with minimum distance value and not already included in SPT (not in
sptSET). The vertex 1 is picked and added to sptSet. So sptSet now becomes {0, 1}. Update
the distance values of adjacent vertices of 1. The distance value of vertex 2 becomes 12,
(See Fig. 3.87).
sptSET). Vertex 7 is picked. So sptSet now becomes {0, 1, 7}. Update the distance values of
adjacent vertices of 7. The distance value of vertex 6 and 8 becomes finite (15 and 9
respectively), (See Fig. 3.88).
Fig. 3.87
Fig. 3.88
sptSET). Vertex 6 is picked. So sptSet now becomes {0, 1, 7, 6}. Update the distance values
of adjacent vertices of 6, (See Fig. 3.89). The distance value of vertex 5 and 8 are updated.
3.78
We repeat the above steps until sptSet doesn’t include all vertices of given graph.
Finally, we get the Shortest Path Tree (SPT) as shown in Fig. 3.90.
Fig. 3.89 Fig. 3.90
Difference between Distance Vector Routing and Link State Routing:
Sr. Distance Vector Link State

No. Routing Routing
1. The distance vector routing The link state routing uses the Shortest
determines the direction (vector) and Path First (SPF) algorithm to create an
distance (such as link cost or number abstract of the exact topology of the
of hops) to any link in the network. entire network.
2. Distance vector routing protocols do A link state routing protocol is like
not have an actual map of the network having a complete map of the network
topology. topology.
3. The distance vector routing algorithm The link state routing algorithm
is a type of routing algorithm that is broadcasts information about the cost of
based on the number of hops in a reaching each of its neighbors to all
route between a source and desti- other routers in the network.
nation computer.
4. Uses Bellman-Ford algorithm. Uses Dijkstra's algorithm.
5. The name 'distance vector' is used In link state routing, each routing node
because the routers exchange vectors makes a connectivity graph for the
containing distance and direction nodes in the network and
information. independently calculates its shortest
path to every other destination in the
network.
6. Less bandwidth is required. High bandwidth is required.
Contd...
3.79
7. Distance vector routing updates full Link state routing updates only the link
routing table. state.
8. Example of distance vector routing Example of link state routing protocols
protocols is RIP. is OSPF.
9. The utilization of CPU and memory in Higher utilization of CPU and memory.
distance vector routing is lower than
the link state routing.
10. Distance vector routing does not have Link state routing works best for
any hierarchical design. hierarchical routing design and in
networks where fast convergence is
crucial.
3.8.3.3 Path Vector Routing Algorithm

• A path vector routing is a more recent concept compared to both a distance vector
routing and the link state routing.
• The path vector routing approach not only exchanges information about the existence
of destination networks but also exchanges the path on how to reach the destination.
• Path information is used to determine the best paths and to prevent routing loops. The
only widely used path vector protocol is BGP.
• Distance vector routing and link state routing are both intra-domain routing protocols.
They can be used inside an autonomous system, but not between autonomous systems.
• These two protocols are not suitable for inter-domain routing mostly because of
scalability. Both of these routing protocols become intractable when the domain of
operation becomes large. Distance vector routing is subject to instability if there are
more than a few hops in the domain of operation.
• Link state routing needs a huge amount of resources to calculate routing tables. It also
creates heavy traffic because of flooding. There is a need for a third routing protocol
which we call path vector routing.
• Path vector routing is used for inter-domain routing with low computational overhead
and support of heterogeneous policies and securities advantages.
• Path vector routing proved to be useful for inter-domain routing. The principle of path
vector routing is similar to that of distance vector routing.
• In path vector routing, we assume that there is one node in each autonomous system
that acts on behalf of the entire autonomous system.
3.80
• Initialization: At the beginning, each speaker node can know only the reach ability of
nodes inside its autonomous system.
Fig. 3.91: Initial Routing Table in Path Vector Routing
• In Fig. 3.91, Node A1 is the speaker node for AS1, B1 for AS2, C1 for AS3, and D1 for
AS4. Node A1 creates an initial table that shows A1 to A5 are located in ASI and can be
reached through it. Node B1 advertises that B1 to B4 are located in AS2 and can be
reached through B1 and so on.
• A path vector protocol is a network routing protocol which maintains the path
information that gets updated dynamically.
• Updates which have looped through the network and returned to the same node are
easily detected and discarded. Border Gateway Protocol (BGP) is an example of a path
vector protocol.
Disadvantages of Path Vector Routing:
1. Lack of Congestion Control: In path vector routing, the routing policies may be
heterogeneous across the ASs. The network traffic or the link congestion may not
be the criterion for path selection in the policies of any AS. Moreover, the path
vector routing protocol converges very quickly and stabilizes. Thus, it may not be
suitable for handling network congestion efficiently.
2. Complex: Path vector routing can be very complex to configure in the network.
3.81
3. Load Balancing: Load balancing between the source and destination can be done
by disseminating packets through each of the alternative paths, multiple paths for
packet dissemination are not selected to support load balancing.
4. Inefficient Load Balancing: The basic path vector routing protocol does not
support load balancing. The path vector table may contain alternative paths to a
destination, but an alternative path is selected generally on the failure of an
existing path.
PRACTICE QUESTIONS
1. Which layer is responsible for the delivery of individual packets from the source to
the destination host?
(a) Physical layer (b) Network layer
(c) Data link layer (d) Transport layer
2. Network layer services include ______.
(a) Routing and Forwarding (b) Packetizing
(c) Flow and Error control (d) All of these
3. Which is a situation in the network layer in which too many datagrams are
present in an area of the Internet?
(a) Congestion (b) Packetizing
(c) Flow and Error control (d) All of these
4. The two congestion control mechanisms are ______.
(a) Open loop (b) Closed loop
5. Which protocol is a set of rules that dictate how data should be delivered over the
Internet
(a) IP (b) UDP
(c) TCP (d) All of these
6. Which is an address used to uniquely identify a device on an IP network?
(a) UDP (b) IP
(c) TCP (d) All of these
7. The parts of an IP are ______.
(a) Network ID (b) Host ID
8. Which is a 32-bits IP address?
(a) IPv4 (b) IPv6
3.82
9. IPv4 addressing uses the concept of classes known as ______ addressing.

(a) Classless (b) Classful
10. In a class A address, the first bit of the first octet is always ‘______’.
(a) 0 (b) 10
(c) 110 (d) 1110
11. Which is a process that extracts the address of the physical network from an IP
address.
(a) Subnetting (b) Supernetting
(c) Masking (d) None of these
12. IP networks can be divided into smaller networks called ______.
(a) Subnets (b) Subnetworks
13. Which is a method for partitioning/dividing a classful IP network into smaller
subnetworks (subnets)?
14. Which is a logical partition of an IP network into multiple, smaller network
segments?
(a) Supernets (b) Subnets
(c) Masks (d) None of these
15. Which allows multiple networks to be specified by one subnet mask?
16. Which is a technology that supports mobile data and applications that are dealing
with wireless connectivity.
(a) Mobile IP (b) Masking IP
(c) Subnet IP (d) None of these
17. Which stores information about mobile nodes visiting its network?
(a) Home Agent (HA) (b) Mobile Node (MN)
(c) Foreign Agent (FA) (d) All of these
18. Which is a network to which the mobile node originally belongs to as per its
assigned IP address (home address)?
(a) Home network (b) Mobile network
(c) Foreign network (d) None of these
3.83
19. Which translates the IP addresses of computers in a local network to a single IP

address?
(a) Mask (b) Supernet
(c) NAT (d) None of these
20. A datagram is a variable length packet which contains two parts namely,
(a) Header (b) Data
21. The three phases of Mobile IP (MIP) includes,
(a) agent discovery (involves the mobile host, the foreign agent and the home
agent)
(b) registration (also involves the mobile host and the two agents)
(c) data transfer (remote host is also involved)
(d) All of these
22. Which is a network layer protocol that enables data communications over a packet
switched network?
(a) UDP (b) IPv6
(c) TCP (d) TCP/IP
23. Which headers allow additional functionality to be implemented in an IPv6
packet?
(a) Forwarding (b) Back wording
(c) Extension (d) None of these
24. Which is the process of establishing the routes that data packets must follow to
reach the destination?
(a) Switching (b) Routing
(c) Forwarding (d) None of these
25. Which is used for routing the packets in routing?
(a) Table (b) NAT
(c) Algorithms (d) None of these
26. Which routing algorithms do not base their routing decisions on measurements or
estimates of the current traffic and topology?
(a) Adaptive (b) Non-adaptive
27. Which algorithms change their routing decisions to reflect changes in the topology,
and usually the traffic as well.
(a) Adaptive (b) Non-adaptive
3.84
28. Which is a routing algorithm in which every router maintains a database with one
entry for each possible destination on the network?
(a) Path Vector (b) Link State
(c) Distance Vector (d) None of these
ANSWERS
1. (b) 2. (d) 3. (a) 4. (c) 5. (a) 6. (b) 7. (c)
8. (a) 9. (b) 10. (a) 11. (c) 12. (c) 13. (a) 14. (b)
15. (b) 16. (a) 17. (c) 18. (a) 19. (c) 20. (c) 21. (d)
22. (b) 23. (c) 24. (b) 25. (c) 26. (b) 27. (a) 28. (c)
1. ______ layer is responsible for routing packets from the source host to the
destination host.
2. ______ control in the network layer is the process of detecting and correcting data
packets that have been corrupted or lost during transmission.
3. The network layer receives the data from the upper layers and creates its own
packets by encapsulating these packets. The process is known as ______.
4. ______ control refers to the techniques used to control or prevent congestion.
5. Congestion control refers to techniques and mechanisms that can either prevent
congestion, before it ______, or remove congestion, after it has ______.
6. The Internet Protocol Address (IP Address) is a unique ______ assigned to every
computing device, such as personal computers, tablets, and smartphones use to
identify itself and communicate with other devices in the IP network.
7. ______ is responsible for packetizing, forwarding and delivery of a packet at the
network layer.
8. Dividing a large block of addresses into several contiguous sub-blocks and
assigning these sub-blocks to different smaller networks is called ______.
9. In ______ addressing, the address space is divided into five classes namely, A, B, C, D
and E.
10. A ______ mask (or number) is used to determine the number of bits used for the
subnet and host portions of the address.
11. When IP is used as a connection less protocol, ______ is based on the destination
address of the IP datagram when the IP is used as a connection oriented protocol,
forwarding is based on the label attached to an IP datagram.
12. Internet Protocol is ______ and unreliable protocol.
3.85
13. Packets in the network (internet) layer are called ______.

14. IPv6 addresses have ______ bits.
15. ______ address in IPv6 identifies a single network interface.
16. A ______ is a networking device that forwards data packets between computer
networks. (#ing)
17. Routers perform the traffic directing functions on the ______.
18. A subnet is a ______ partition of an IP network into multiple, smaller network
segments.
19. A routing ______ is a method for determining the routing of packets in a node. For
each node of a network, the algorithm determines a routing table, which in each
destination, matches an output line.
20. ______ vector routing algorithms require that each node exchanges
information between neighbors, that is to say between nodes directly connected.
21. A routing ______ is a grouping of information stored on a networked computer or
network router that includes a list of routes to various network destinations.
22. The ______ state routing algorithm is used to find the shortest path from one node
to every other node in the network.
23. In ______ addressing, there are no classes, but the addresses are still granted in
blocks.
ANSWERS
1. Network 2. Error 3. packetizing 4. Congestion
5. happens, happened 6. number 7. IPv4 8. subnetting
9. classful 10. subnet 11. forwarding 12. connectionless
13. datagrams 14. 128 15. Unicast 16. router
17. Internet 18. logical 19. algorithm 20. Distance
21. table 22. link 23. classless

1. With the help of forwarding, data packets are transferred from one place to
another in the network.
2. Error occurs when the number of datagrams sent by source is beyond the capacity
of the network.
3.86
4. Open loop congestion control policies are applied to prevent congestion after it
happens.
5. In general, we can divide congestion control mechanisms into two broad
categories namely, open-loop congestion control (prevention) and closed-loop
congestion control (removal).
6. Backpressure is a node-to-node congestion control technique that propagate in the
opposite direction of data flow.
7. Choke packet technique is applicable to both virtual networks as well as datagram
subnets. A choke packet is a packet sent by a node to the source to inform it of
congestion.
8. When too many packets are present in a subnet (or part of a subnet), performance
degrades. Packets sent are not equal to the packets received. This situation is called
flow control.
9. IPv4 is a 32-bits IP address.
32
10. The address space of IPv4 is 2 or 4,294,967,296.
11. A mask used to determine what subnet an IP address belongs to.
12. In a class C address, the first octet would always start with ‘10’. Thus, class C
addresses range from 192.0.0.0 to 223.255.255.255.
13. Network Address Translation (NAT) is a process in which one or more local IP
address is translated into one or more Global IP address and vice versa in order to
provide Internet access to the local hosts.
14. The supernets are created through the use of subnet masks.
15. The subset mask identifies which hits in the IP address are to be used to represent
the network subnet portion of an IP address.
16. Gateways work on a network layer and provide mechanism to route data to its
destination.
17. Mobile IP has two addresses for a mobile host (home address: permanent) it
associates the host with its home network and Care-of address Changes as the
mobile host moves from one network to another and it is associated with the
foreign network.
18. Internet Protocol version 6 (IPv6) is the most recent version of the Internet
Protocol (IP).
19. IPv6 is the next generation Internet Protocol (IP) standard intended to eventually
replace IPv4.
20. Subnetting is the process of dividing (partitioning) a network into several smaller
networks (subnets).
3.87
21. A router is a device that forwards packets between networks by processing the
routing information included in the packet.
22. RIP (Routing Information Protocol) is the best example of a protocol using distance
vector.
23. Subnetting, also called Classless Inter-Domain Routing (CIDR), is a way to aggregate
multiple Internet addresses of the same class.
24. CIDR notation specifies the number of bits set to a 1 that make up the subnet mask.
ANSWERS
1. (T) 2. (F) 3. (T) 4. (F) 5. (T) 6. (T)
7. (T) 8. (F) 9. (T) 10. (T) 11. (T) 12. (F)
13. (T) 14. (F) 15. (T) 16. (F) 17. (T) 18. (T)
19. (T) 20. (T) 21. (T) 22. (T) 23. (F) 24. (T)
1. What are the services provided by the network layer?
2. List functions of the network layer.
3. What is flow and error control?
4. Define congestion. When it occurs.
5. List types of congestion control mechanisms.
6. What is an IP address?
7. List classes for classful IP addressing.
8. Define masking.
9. What is meant by subnetting?
10. Define supernetting.
11. What is meant by classless addressing/
12. What is NAT?
13. List network layer protocols.
14. What is fragmentation?
15. What is Mobile IP?
16. List three phases of Mobile IP.
17. What is IPv6?
18. List address for IPv6.
19. What is an extension header?
20. Define routing.
3.88
21. Give role of routing algorithm in routing.

22. List various routing algorithms.
1. With the help of diagrams describe congestion control in the network layer.
2. Write a short note on: IPv4 addressing.
3. With the help of diagram and example describe format of an IP address.
4. Give address space for IPv4 and IPv6 addresses.
5. Describe classful addressing with diagrams.
6. Explain the term masking and subnet masking with example.
7. Explain supernetting with example.
8. Define NAT. How does it work? Explain with an example.
9. Describe forwarding IP packets based on destination address and based on label.
10. Describe IP protocol with datagram format.
11. Explain the term fragmentation with its related fields.
12. With the help of a diagram describe Mobile IP (MIP) architecture.
13. Define the terms HA and FA in MIP.
14. Explain phases of Mobile IP diagrammatically.
15. Describe IPv6 protocol with packer format.
16. Write a short note on: Extension header.
17. Differentiate between IPv4 and IPv6.
18. Define routing and routing algorithm.
19. With the help of example describe distance vector routing algorithm.
20. Describe link state routing algorithm with example.

April 2016
1. Find out the class, Net ID and Host ID of IP address 126.47.50.23. [1 M]
2. Explain classful addressing in detail. [5 M]
3. List different task performed by the network layer. [2 M]
April 2017
1. Find out class, netid and hostid of IP address 126.25.21.1. [4 M]
3.89
October 2017
1. What is fragmentation? [1 M]
2. State the class of the IP address 128.89.0.26. [1 M]
3. Explain open and closed loop congestion control mechanisms. [5 M]
Ans. Refer to Section 3.2, Points (1) and (2).
April 2018
1. Identify the class of the IP addresses 192.168.60.12 and 10.11.1.1. [1 M]
2. Define subnetting. [1 M]
3. What is routing? Explain the desirable characteristics. [5 M]
October 2018
1. What is fragmentation? [1 M]
2. Define routing. [1 M]
3. What is congestion? [5 M]
4. What are the services provided by the network layer? [5 M]
5. List the congestion control policies used at the network layer. [2 M]
April 2019
1. Convert dotted decimal IP address to binary address 255.255.0.0. [1 M]
2. What is congestion? [1 M]
3. Explain Host id and Net id of IP address classes. [4 M]
3.90
CHAPTER
4
Transport Layer
Objectives…
To learn Transport Layer with its Services
To study UDP and TCP Protocols
4.0 INTRODUCTION
• The transport layer is responsible for providing services to the application layer, it
receives services from the network layer.
• Transport layer, (layer 4) in the TCP/IP model is responsible for process-to-process
communication, flow control, congestion control and so on.
• The original TCP/IP protocol suite specifies two protocols for the transport layer
namely, UDP and TCP.
• A new transport layer protocol SCTP is recently developed for signaling message
transport over IP networks.
4.1 TRANSPORT LAYER SERVICES

• The transport layer provides communication services between the computers
connected in the network.
• In this section, we discuss the services that can be provided by a transport layer like
process-to-process communication, addressing, encapsulation and decapsulation,
multiplexing and demultiplexing, flow control, congestion control and so on.
4.1.1 Process-to-Process Communication

• The first duty of a transport layer is to provide process-to-process communication. A
process is an application program running on the host and uses the services of the
transport layer.
• Before we discuss how process-to-process communication can be accomplished, we
need to understand the difference between host-to-host communication and process-
to-process communication.
• The data link layer is responsible for delivery of frames between two neighboring
nodes over a link. This is called node-to-node communication.
4.1
Computer Networks – I (S.Y.B.Sc. Comp. Science) Transport Layer
• The network layer is responsible for delivery of datagram between two hosts. This is
called host-to-host communication.
• Real communication takes place between two processes (application programs) in a
network. This is called process-to process communication.
• The transport layer is responsible for process-to-process communication, the delivery
of a packet, part of a message, from one process to another.
• A network layer protocol can deliver the message only to the destination computer.
However, this is an incomplete delivery.
• The message still needs to be handed to the correct process. This is where a transport
layer protocol takes over.
• A transport layer protocol is responsible for delivery of the message to the appropriate
process.
• Fig. 4.1 shows the domains of a network layer and a transport layer.
Processes Processes
Node to node: Data link layer
Host to host: Network layer
Process to process: Transport layer
Internet
Node to Node to Node to Node to

node node Node to node node node
Host to host
Process to process
Fig. 4.1: Network Layer versus Transport Layer
4.1.2 Addressing: Port Numbers

• The most common method to achieve process to process communication is through
the client/server paradigm.
• A process on the local host, called a client, needs services from a process on the remote
host called a server.
• The client and server machines can run several applications programs
simultaneously.
• Both processes (client and server) have the same name. For example, to get the day
and time from a remote machine, we need a day time client process running on the
local host and day time server process running on a remote machine.
4.2
• However, Operating Systems (OS) today support both multiuser and multi-
programming environments.
• A remote computer can run several server programs at the same time, just as several
local computers can run one or more client programs at the same time.
• For communication, we must define the local host, local process, remote host, and
remote process.
• The local host and the remote host are defined using IP addresses. To define the
processes, we need second identifiers, called port numbers.
• In the TCP/IP protocol suite, the port numbers are integers between 0 and 65,535 (16
bits).
• Port numbers are 16 bit long that help identify which process is sending or receiving
data on a host.
• The client program defines itself with a port number, called the ephemeral port
number. The word ‘ephemeral’ means ‘short lived’ and is used because the life of a
client is normally short.
• The server process must also define itself with a port number. This port number,
however, cannot be chosen randomly.
• If the computer at the server site runs a server process and assigns a random number
as the port number, the process at the client site that wants to access that server and
use its services will not know the port number.
• TCP/IP has decided to use universal port numbers for servers; these are called well-
known port numbers.
• Every client process knows the well-known port number of the corresponding server
process.
• For example, while the Daytime client process use an ephemeral (temporary) port
number 52,000 to identify itself, the Daytime server process must use the well-known
(permanent) port number 15 (See Fig. 4.2).
Daytime client Daytime server
52,000 15
Transport layer Transport layer
Data 15 52,000
15 52,000 Data
Fig. 4.2: Port Numbers
4.3
Addressing:
• To deliver the data, we need an address. At the data link layer, we need a physical
(MAC) address. A frame in the data link layer needs a destination MAC address for
delivery and source MAC address for reply.
• At the network layer, we need an IP address. A datagram in the network layer needs a
destination IP address for delivery and a source IP address for the destination's reply.
• At the transport layer, we need a transport layer address called port address/number
to choose among multiple processes running on the destination host.
• The destination port number is needed for delivery; the source port number is needed
for the reply.
• The IP addresses and port numbers play different roles in selecting the final
destination of data. The role of destination IP address is to define the host among the
different hosts.
• Once the host has been selected, the port number starts, the port number defines one
of the processes on this particular host. This is shown in Fig. 4.3.
15
Port number
15
selects the process
190.15.25.5
IP header IP address
190.15.25.5
selects the host
Transport layer 15
header
Fig. 4.3: The Role of Destination IP Address

IANA Ranges:
• The IANA (Internet Assigned Number Authority) has divided the port numbers into
three ranges as shown in Fig. 4.4.
4.4
Registered
0 1023 49,152 65,535
1024 49,151
Well known Dynamic (ephemeral)
Fig. 4.4: IANA Ranges

• Fig. 4.4 shows follwing IANA Ragnes:
1. Well Known Ports: The ports from 0 to 1023 are assigned and controlled by IANA.
These are well known ports.
2. Registered Ports: The ports ranging from 1024 to 49151 are not assigned or
controlled by IANA. They can be registered with IANA to avoid duplication.
3. Dynamic Ports (Ephemeral Ports): The ports ranging from 49,152 to 65,535 are
neither controlled nor registered. They can be used by any process. These are
ephemeral ports.
Socket Addresses:
• Transport layer provides process-to-process communication which needs two
identifiers, IP address and port number at each end to make a connection.
• The combination of an IP address and a port number is called a socket address.
• The client socket address shows the client process uniquely and the server socket
address shows the server process uniquely.
Fig. 4.5: Socket Address
4.1.3 Encapsulation and Decapsulation

• To send a message from one process to another, the transport layer encapsulates and
decapsulates messages as shown in Fig. 4.6.
• Encapsulation happens at the sender site. When a process has a message to send, it
passes the message to the transport layer along with a pair of socket addresses and
some other pieces of information that depends on the transport layer protocol.
• The transport layer receives the data and adds the transport-layer header. The packets
at the transport layers in the Internet are called user datagrams, segments, or packets.
• Decapsulation happens at the receiver site. When the message arrives at the
destination transport layer, the header is dropped and the transport layer delivers the
message to the process running at the application layer.
4.5
• The sender socket address is passed to the process in case it needs to respond to the
message received.
(a) Encapsulation (b) Decapsulation
Fig. 4.6: Encapsulation and Decapsulation in Transport Layer
4.1.4 Multiplexing and Demultiplexing (April 17)
• Multiplexing and demultiplexing are the two very important services that are
performed by the transport layer.
• The transport layer at the source performs multiplexing while the transport layer at
the destination performs demultiplexing as shown in Fig. 4.7.
• Whenever an entity accepts items from more than one source, it is referred to as
multiplexing (many to one); whenever an entity delivers items to more than one
source, it is referred to as demultiplexing (one to many).
• Fig. 4.7 shows communication between a client and two servers. Three client
processes are running at the client site namely, P1, P2, and P3.
• The processes P1 and P3 need to send requests to the corresponding server process
running in a server.
• The client process P2 needs to send a request to the corresponding server process
running at another server.
• The transport layer at the client site accepts three messages from the three processes
and creates three packets. It acts as a multiplexer.
• The packets 1 and 3 use the same logical channel to reach the transport layer of the
first server.
• When they arrive at the server, the transport layer does the job of a multiplexer and
distributes the messages to two different processes.
• The transport layer at the second server receives packet 2 and delivers it to the
corresponding process.
4.6
Fig. 4.7: Multiplexing and Demultiplexing
4.1.5 Flow Control

• The transport layer provides a flow control mechanism between the adjacent layers of
the TCP/IP model.
• TCP also prevents data loss due to a fast sender and slow receiver by imposing some
flow control techniques.
• In communication at the transport layer, we are dealing with four entities namely,
sender process, sender transport layer, receiver transport layer, and receiver process.
• The sending process at the application layer is only a producer. It produces message
chunks and pushes them to the transport layer.
• The sending transport layer has a double role, it is both a consumer and the producer.
It consumes the messages pushed by the producer.
4.7
• It encapsulates the messages in packets and pushes them to the receiving transport
layer.
• The receiving transport layer has also a double role, it is the consumer for the packets
received from the sender. It is also a producer; it needs to decapsulate the messages
and deliver them to the application layer.
• The last delivery, however, is normally a pulling delivery; the transport layer waits
until the application-layer process asks for messages.
• Fig. 4.8 shows that we need at least two cases of flow control namely, from the sending
transport layer to the sending application layer and from the receiving transport layer
to the sending transport layer.
Fig. 4.8: Flow Control at the Transport Layer
4.1.5.1 Pushing or Pulling

• Delivery of items from a producer to a consumer can occur in one of the two ways
namely, pushing or pulling.
1. If the sender delivers items whenever they are produced without the prior request
from the consumer the delivery is referred to as pushing.
2. If the producer delivers the items after the consumer has requested them, the
delivery is referred to as pulling.
• Fig. 4.9 shows above two types of delivery (pushing and pulling).
(a) Pushing (b) Pulling

Fig. 4.9: Pushing and Pulling
4.8
• When the producer pushes the items, the consumer may be overwhelmed and there is
a need for flow control, in the opposite direction, to prevent the discarding of the
items.
• In other words, the consumer needs to warn the producer to stop the delivery and to
inform it when it is ready again to receive the items.
• When the consumer pulls the items, it requests them when it is ready. In this case,
there is no need for flow control.
4.1.5.2 Buffers
• Although flow control in the transport layer can be implemented in several ways, one
of the solutions is normally to use two buffers. One at the sending transport layer and
the other at the receiving transport layer.
• A buffer is a set of memory locations that can hold packets at the sender and receiver.
The flow control communication can occur by sending signals from the consumer to
producer.
• When the buffer of the sending transport layer is full, it informs the application layer
to stop passing chunks of messages; when there are some vacancies, it informs the
application layer that it can pass message chunks again.
• When the buffer of the receiving transport layer is full, it informs the sending
transport layer to stop sending packets.
• When there are some vacancies, it informs the sending transport layer that it can send
messages again.
• For example: The consumers communicate with the producers on two occasions such
as, when the buffer is full and when there are vacancies. If the two parties use a buffer
of only one slot, the communication can be easier. Assume that each transport layer
uses one single memory location to hold a packet. When this single slot in the sending
transport layer is empty, the sending transport layer sends a note to the application
layer to send its next chunk; when this single slot in the receiving transport layer is
empty, it sends an acknowledgment to the sending transport layer to send its next
packet. As we will see later, this type of flow control, using a single-slot buffer at the
sender and the receiver, is inefficient.
4.1.6 Error Control

• The error control service in the transport layer, observes that the data delivered to the
receiver is error free and reliable.
• Reliability can be achieved to add error control service to the transport layer. Error
control at the transport layer is responsible to:
1. Detect and discard corrupted packets.
4.9
2. Keep track of lost and discarded packets and resend them.

3. Recognize duplicate packets and discard them.
4. Buffer out-of-order packets until the missing packets arrive.
• Error control involves only the sending and receiving transport layers. We are
assuming that the message chunks exchanged between the application
and transport layers are error free.
• Fig. 4.10 shows the error control between the sending and receiving transport layer.
• As with the case of flow control, the receiving transport layer manages error control,
most of the time, by informing the sending transport layer about the problems.
Fig. 4.10: Error Control at the Transport Layer
4.1.6.1 Sequence Numbers

• Error control mechanism requires that the sending transport layer knows which
packet is to be resent and the receiving transport layer knows which packet is a
duplicate, or which packet has arrived out of order.
• This can be done if the packets are numbered. We can add a field to the transport
layer packet to hold the sequence number of the packets.
• When a packet is corrupted or lost, the receiving transport layer can somehow inform
the sending transport layer to resend that packet using the sequence number.
• The receiving transport layer can also detect duplicate packets if two received packets
have the same sequence number. The out-of-order packets can be recognized by
observing gaps in the sequence numbers.
• Packets are numbered sequentially. However, because we need to include the
sequence number of each packet in the header, we need to set a limit.
• If the header of the packet allows m bits for the sequence number, the sequence
m
numbers range from 0 to 2 − 1.
• For example, if m is 4, the only sequence numbers are 0 through 15, inclusive.
However, we can wrap around the sequence. So the sequence numbers in this case
are:
0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, ...
m
• In other words, the sequence numbers are modulo 2 .
4.10
4.1.6.2 Acknowledgements
• In error control, we can use both positive and negative signals as error control. The
receiver side can send an acknowledgement (ACK) for each or a collection of packets
that have arrived safe and sound.
• The receiver can simply discard the corrupted packets. The sender can detect lost
packets if it uses a timer.
• When a packet is sent, the sender starts a timer; when the timer expires, if an ACK
does not arrive before the timer expires, the sender resends the packet.
• Duplicate packets can be silently discarded by the receiver. Out-of-order packets can
be either discarded (to be treated as lost packets by the sender), or stored until the
missing ones arrive.
4.1.6.3 Sliding Window

m
• Since the sequence numbers used modulo 2 , a circle can represent the sequence
m
number from 0 to 2 − 1 as shown in Fig. 4.11.
• The buffer is represented as a set of slices, called the sliding window, that occupy part
of the circle at any time.
• At the sender site, when a packet is sent, the corresponding slice is marked. When all
the slices are marked, it means that the buffer is full and no further messages can be
accepted from the application layer.
• When an acknowledgment arrives, the corresponding slice is unmarked. If some
consecutive slices from the beginning of the window are unmarked, the window slides
over the range of the corresponding sequence number to allow more free slices at the
end of the window.
(a) Four Packets have been Sent (b) Five Packets have been Sent
4.11
(c) Seven Packets have been Sent (d) Packet 0 has been Acknowledged
Window is Full Window Slides
Fig. 4.11: Sliding Window in Circular Format
• Fig. 4.11 shows the sliding window at the sender. The sequence number are modulo 16
(m = 4) and the size of the window is 7.
• Note that the sliding window is just an abstraction: the actual situation uses computer
variables to hold the sequence number of the next packet to be sent and the last packet
sent.
• Number of protocols show the sliding window using linear representation. The idea is
the same, but it normally takes less space on paper.
• Fig. 4.12 shows linear representation of the sliding window.
(a) Four Packets have been Sent (b) Five Packets have been Sent
(c) Seven Packets have been Sent (d) Packet 0 have been Acknowledged
Window is Full and Window Slides
Fig. 4.12: Sliding Window in Linear Format
• Both representations (circular and linear) of the sliding window, tell us the same
thing. If we take both sides of each part in Fig. 4.11 and bend them up, we can make
the same part in Fig. 4.12.
4.12
4.1.7 Congestion Control

• Congestion is a situation in which too many sources over a network attempt to send
data and the router buffers start overflowing due to which loss of packets occur.
• Congestion in a network may occur if the load on the network - the number of packets
sent to the network - is greater than the capacity of the network - the number of
packets a network can handle.
• Congestion control refers to the mechanisms and techniques to control the congestion
and keep the load below the capacity.
• Congestion happens in any system that involves waiting. Congestion in a network or
internetwork occurs because routers and switches have queues -buffers that hold the
packets before and after processing.
• The packet is put in the appropriate output queue and waits its turn to be sent. These
queues are finite, so it is possible for more packets to arrive at a router than the router
can buffer.
• Congestion control refers to techniques and mechanisms that can either prevent
• Congestion control uses open loop congestion control to prevent the congestion and
closed loop congestion control to remove the congestion in a network once it occurs.
CONNECTIONLESS AND CONNECTION ORIENTED

4.2
SERVICES
• A transport layer provides two types of services namely, connectionless and
connection-oriented.
1. Connectionless Service:
• In a connectionless service, the packets are sent from one machine to another without
connection establishment or connection release.
• The packets may arrive without order or they may be lost or delayed. Packets are not
numbered. No acknowledgement is processed.
• In a connectionless service, the source process (application program) needs to divide
its message into chunks of data of the size acceptable by the transport layer and
deliver them to the transport layer one by one.
• The transport layer treats each chunk as a single unit without any relation between
the chunks. When a chunk arrives from the application layer, the transport layer
encapsulates it in a packet and sends it.
• To show the independence of packets, assume that a client process has three chunks of
messages to send a server process.
4.13
• The chunks are handed over to the connectionless transport protocol in order.
• However, since there is no dependency between the packets at the transport layer, the
packets may arrive out of order at the destination and will be delivered out of order to
the server process.
• In Fig. 4.13, we have shown the movement of packets using a timeline, but we have
assumed that the delivery of the process to the transport layer and vice versa are
instantaneous.
• The Fig. 4.13, shows that at the client site, the three chunks of messages are delivered
to the client transport layer in order (1, 2, and 3).
• Because of the extra delay in transportation of the second packet, the delivery of
messages at the server is not in order (1, 3, 2).
• If these three chunks (1, 2, and 3) of data belong to the same message, the server
process may have received a strange message.
Fig. 4.13: Connectionless Service
• UDP is a transport layer’s connectionless protocol.

2. Connection Oriented Service:
• In a connection oriented service, connection is established first and then data are
transferred in between sender and receiver. After the end of data transfer, connection
is released.
• In a connection-oriented service, the client and the server first need to establish a
connection between themselves.
4.14
• The data exchange can only happen after the connection establishment. After data
exchange, the connection needs to be teared down as shown in Fig. 4.14.
• As we mentioned before, the connection-oriented service at the transport layer is
different from the same service at the network layer.
• In the network layer, connection-oriented service means a coordination between the
two end hosts and all the routers in between.
• At the transport layer, connection-oriented service involves only the two hosts; the
service is end to end.
• This means that we should be able to make a connection-oriented protocol over either
a connectionless or connection-oriented protocol.
Fig. 4.14: Connection-oriented Service

4.15
• Fig. 4.14 shows the connection establishment, data transfer, and teardown phases in a
connection-oriented service at the transport layer.
• Note that most protocols combine the third and fourth packets in the connection
establishment phase into one packet.
• TCP and SCTP, these two transport layer protocols are connection oriented.
4.3 TRANSPORT LAYER PROTOCOLS (Oct. 18)
• In the Internet protocol suite, the transport layer supports three protocols TCP, SCTP
and UDP. Out of which TCP and SCTP are connection oriented and reliable and UDP is
connectionless and unreliable.
• Fig. 4.15 shows the position of UDP, TCP and SCTP in the TCP/IP protocol suite.
Fig. 4.15
4.3.1 User Datagram Protocol (UDP) (April 16, 18)

• The User Datagram Protocol (UDP) is the simplest Transport Layer communication
protocol in the TCP/IP protocol suite and serves as the intermediary between the
application programs and the network operations.
• The UDP was designed by David P. Reed in 1980 and formally defined in RFC 768
standard.
• UDP is a connectionless, unreliable Transport Layer protocol.
• UDP does not add anything to the services of IP except to provide process-to-process
communication instead of host-to-host communication. Also, it performs very limited
error checking.
• UDP is powerless protocol. But it is a very simple protocol, using minimum overhead.
If a process wants to send a small message and does not care much about reliability, it
can use UDP.
4.16
• UDP is stateless protocol. It is a suitable protocol for streaming applications such as

VoIP, multimedia streaming.
Features of UDP:
1. Connectionless Service: UDP provides connectionless service. In UDP each packet
is independent from other packets sent by the same application.
2. Lack of Congestion Control: UDP does not provide congestion control and it does
not create additional traffic in an error prone network.
3. Lack of Error Control: UDP does not provide error control. So UDP provides
unreliable service.
4. Transaction-oriented: UDP is transaction-oriented, suitable for simple query-
response protocols such as the Domain Name System (DNS).
5. Datagram: UDP provides datagrams, suitable for modeling other protocols such as
IP tunneling or Remote Procedure Call (RPC) and the Network File System (NFS).
An UDP datagram is used in Network File System (NFS), DNS, SNMP, TFTP etc.
6. Simple: UDP is a simple, datagram-oriented, transport-layer protocol. UDP is
simple, suitable for bootstrapping or other purposes without a full protocol stack,
such as the DHCP and Trivial File Transfer Protocol (TFTP).
7. Stateless: UDP is stateless, suitable for very large numbers of clients, such as in
streaming media applications such as IPTV.
8. Lack of Retransmission Delays: The lack of retransmission delays makes UDP
suitable for real-time applications such as Voice over IP, online games, and many
protocols using Real Time Streaming Protocol.
9. Support Multicast: Because it supports multicast, it is suitable for broadcast
information such as in many kinds of service discovery and shared information
such as Precision Time Protocol (PTP) and Routing Information Protocol (RIP).
10. Faster in Data Transfer: UDP is a lightweight protocol for faster and simpler data
transmissions.
11. Queuing: UDP is simple and suitable for query based communications. The
Queues are associated with the ports in UDP.
12. Low Overhead: UDP is designed to provide application processes with the ability
to transfer data with a minimal overhead.
13. Port: UDP uses that concept of port, which allows to distinguish the different
applications running on a machine. Besides the datagram and its data, a UDP
message contains a source port number and destination port number.
14. No Acknowledgment/Not Reliable: In UDP data are transmitted with no
acknowledgment of whether it is received or not. UDP is thus not as reliable as
TCP. UDP does not guarantee ordered delivery of data.
4.17
Uses of UDP: (Oct. 17)

1. UDP is suitable for a process that requires simple request response communication
with little concern for flow control and error control.
2. UDP is suitable for a process having inbuilt error control and flow control
mechanisms, for example, TFTP (Trivial File Transfer Protocol).
3. UDP is used for route updating protocols such as RIP (Routing Information
Protocol).
4. UDP is suitable for multicasting, TCP not.
5. UDP is used for management processes such as SNMP (Simple Network
Management Protocol).
4.3.2 Datagram Format (April 16, 19)
• UDP packets, called as user datagrams, have a fixed-size header of 8 bytes.

• Fig. 4.16 shows the format of a user datagram. Fig 4.17 shows header format of UDP.
8 to 65,535 bytes
8 bytes
Header Data
Fig. 4.16: UDP User Datagram
0 16 31
Source Port Number Destination Port Number
Total Length Checksum
Fig. 4.17: Header Format of UDP
• UDP header contains four main parameters:

1. Source Port Number: This 16-bits field is used by the process running on the
source host which wants to make communication. Port numbers can range from 0
to 65,535. If a client is sending a request, generally the port number is an
ephemeral port number. If the server is sending a response port number is a well
known port number.
2. Destination Port: This 16-bits is used by the process running on the destination
host. If the destination host is a server, the port number is a well known port
number. If the destination host is a client, the port number is an ephemeral port
number.
4.18
3. Length: This 16-bits field defines the total length of the user datagram, header plus
data. This field is actually not necessary, because UDP is encapsulated in IP. IP has
total length and header length fields.
So, UDP length = IP length – IP header’s length
4. Checksum: This 16-bits field is used to detect errors over the entire user datagram.
4.3.3 UDP Services

• The general UDP services are Process to Process Communication, Connectionless
Services, Flow Control, Error Control, Congestion Control, Encapsulation and
Decapsulation, Queuing, Multiplexing and Demultiplexing.
1. Process to Process Communication:
• UDP provides a process to process communication using sockets, a combination of IP
addresses and port numbers.
• Several port numbers used by UDP are shown below:
Port No Protocol Description
7 Echo Resends a received datagram back to the sender.
9 Discard Discards a received datagram.

11 Users Shows active users.
13 Daytime Gives the date and time.
17 Quote Gives a quote of the day.

19 Chargen Gives a string of characters.
53 Nameserver Shows Domain Name Service (DNS).
67 BOOTPs Server port to download bootstrap information.

68 BOOTCPc Client port to download bootstrap information.
69 TFTP Trivial File Transfer Protocol (TFTP).
111 RPC Remote Procedure Call.

123 NTP Network Time Protocol.
161 SNMP Simple Network Management Protocol.
162 SNMP Simple Network Management Protocol.

2. Connectionless Services:
• UDP provides a connectionless service. This means that each user datagram sent by
UDP is an independent datagram.
4.19
• In UDP there is no relationship between the different user datagrams even if they are
coming from the same source process and going to the same destination program.
• The user datagrams are not numbered. There is no connection establishment and no
connection termination as is the case for TCP.
• Each user datagram can travel on a different path. One of the ramifications of being
connectionless is that the process that uses UDP cannot send a stream of data to UDP
and expect UDP to chop them into different related user datagrams.
• Instead each request must be small enough to fit into one user datagram. Only those
processes sending short messages, messages less than 65,507 bytes (65,535 minus 8
bytes for the UDP header and minus 20 bytes for the IP header), can use UDP.
3. Flow Control:
• UDP is a very simple protocol. There is no flow control, and hence no window
mechanism. The receiver may overflow with incoming messages.
• The lack of flow control means that the process using UDP should provide for this
service, if needed.
4. Error Control:
• There is no error control mechanism in UDP except for the checksum.
• Sender does not know if a message has been lost or duplicated. When the receiver
detects an error through the checksum, the user datagram is silently discarded.
• The lack of error control means that the process using UDP should provide for this
service if needed.
Checksum:
• UDP provides checksums for data integrity, and port numbers for addressing different
functions at the source and destination of the datagram.
• The UDP checksum calculation is different from IP. UDP’s checksum includes three
sections namely, a pseudo header, the UDP header and the data coming from the
application layer.
• The pseudo header is the part of the header of the IP packet in which the user
datagram is to be encapsulated with some fields filled with 0s.
• If the checksum does not include the pseudo header, a user datagram may arrive safe
and sound. If the IP header is corrupted, it may be delivered to the wrong host.
• The protocol field is added to confirm that the packet belongs to UDP. The value of
protocol for UDP is 17. If this value is changed during transmission, the checksum
calculation at the receiver will detect it and UDP drops the packet.
4.20
Fig. 4.18: Pseudo Header for Checksum Calculation
• The calculation of checksum and its inclusion in a user datagram are optional. If
checksum is not calculated, the field is filled with 1s.
Example: Fig. 4.19 shows the checksum calculation for a very small user datagram
with only 7 bytes of data. Since the data is odd, padding is added for checksum
calculation. The pseudo header as well as the padding will be dropped when the user
datagram is delivered to IP.
Fig. 4.19: Checksum Calculation of a Simple UDP User Datagram
4.21
5. Multiplexing and Demultiplexing:

• In a host running a TCP/IP protocol suite, there is only one UDP but possibly several
processes that may want to use the services of UDP. To handle this situation, UDP
multiplexers and demultiplexers (See Fig. 4.20).
Processes Processes
Multiplexer Demultiplexer
IP IP
Fig. 4.20: Multiplexing and Demultiplexing

• UDP uses multiplexing to handle outgoing user datagrams from multiple processes on
one host. UDP uses demultiplexing to handle incoming user datagrams that go to
different processes on the same host.
• Multiplexing: At the sender site, there may be several processes that need to send
user datagrams. However, there is only one UDP. This is a many-to-one relationship
and requires multiplexing. UDP accepts messages from different processes,
differentiated by their assigned port numbers. After adding the header, UDP passes
the user datagram to IP.
• Demultiplexing: At the receiver site, there is only one UDP. However, we may have
many processes that can receive user datagrams. This is a one-to-many relationship
and requires demultiplexing. UDP receives user datagrams from IP. After error
checking and dropping of the header, UDP delivers each message to the appropriate
process based on the port numbers.
6. Encapsulation and Decapsulation:
• To send a message from one process to another, the UDP protocol encapsulates and
decapsulates messages as shown in Fig. 4.21.
Encapsulation in UDP:
• When a process has a message to send through UDP, it passes the message to UDP
along with a pair of socket addresses and the length of data.
• UDP receives the data and adds the UDP header. UDP then passes the user datagram to
IP with the socket addresses.
4.22
• IP adds its own header, using the value 17 in the protocol field, indicating that the data
has come from the UDP protocol.
• The IP datagram is then passed to the data link layer. The data link layer receives the
IP datagram, adds its own header (and possibly a trailer), and passes it to the physical
layer.
• The physical layer encodes the bits into electrical or optical signals and sends it to the
remote machine.
Decapsulation in UDP:
• When the message arrives at the destination host, the physical layer decodes the
signals into bits and passes it to the data link layer.
• The data link layer uses the header (and the trailer) to check the data. If there is no
error, the header and trailer are dropped and the datagram is passed to IP.
• The IP software does its own checking. If there is no error, the header is dropped and
the user datagram is passed to UDP with the sender and receiver IP addresses.
• UDP uses the checksum to check the entire user datagram. If there is no error, the
header is dropped and the application data along with the sender socket address is
passed to the process.
• The sender socket address is passed to the process in case it needs to respond to the
message received.
(a) Encapsulation (b) Decapsulation
Fig. 4.21: Encapsulation and Decapsulation in UDP
4.23
7. Queuing:
• In UDP, queues are associated with ports as shown in Fig. 4.22.
• When a process starts, at the client site, it requests a port number from the Operating
System (OS). Some implementations create both an incoming and an outgoing queue
associated with each process while other implementations create only an incoming
queue associated with each process.
• Even if a process wants to communicate with multiple processes, it obtains only one
port number and eventually one outgoing and one incoming queue.
• The queues opened by the client are, in most cases, identified by ephemeral port
numbers. The queues function as long as the process is running.
• When the process terminates, the queues are destroyed. The client process can send
messages to the outgoing queue by using the source port number specified in the
request.
• UDP removes the messages one by one, and after adding the UDP header, delivers
them to IP. An outgoing queue can overflow.
• If this happens, the operating system can ask the client process to wait before sending
any more messages.
• When a message arrives for a client, UDP checks to see if an incoming queue has been
created for the port number specified in the destination port number field of the user
datagram.
• If there is such a queue, UDP sends the received user datagram to the end of the
queue. If there is no such queue, UDP discards the user datagram and asks the ICMP
protocol to send a port unreachable message to the server.
• All of the incoming messages for one particular client program, whether coming from
the same or a different server, are sent to the same queue. An incoming queue can
overflow.
• If this happens, UDP drops the user datagram and asks for a port unreachable message
to be sent to the server.
• At the server site, the mechanism of creating queues is different. In its simplest form, a
server asks for incoming and outgoing queues using its well-known port when it starts
running. The queues remain open as long as the server is running.
• When a message arrives for a server, UDP checks to see if an incoming queue has been
created for the port number specified in the destination port number field of the user
datagram.
• If there is such a queue, UDP sends the received user datagram to the end of the
queue. If there is no such queue, UDP discards the user datagram and asks the ICMP
protocol to send a port unreachable message to the client.
4.24
• All of the incoming messages for one particular server, whether coming from the same
or a different client, are sent to the same queue.
• An incoming queue can overflow. If this happens, UDP drops the user datagram and
asks for a port unreachable message to be sent to the client.
• When a server wants to respond to a client, it sends messages to the outgoing queue
using the source port number specified in the request.
• UDP removes the messages one by one, and, after adding the UDP header, delivers
them to IP.
• An outgoing queue can overflow. If this happens, the operating system asks the server
to wait before sending any more messages.
Fig. 4.22: Queuing in UDP
• UDP does not provide congestion control because it is a connectionless protocol. UDP
assumes that the packets sent are small and sporadic, and cannot create congestion in
the network.
• This UDP assumption may or may not be true today when UDP is used for real-time
transfer of audio and video.
4.4 TRANSMISSION CONTROL PROTOCOL (TCP)

• The TCP is one of the most important protocols of the Internet Protocols suite. TCP is a
reliable and connection oriented protocol.
• TCP is the most widely used protocol for data transmission in communication
networks such as the Internet.
• TCP lies between the application layer and the network layer, and serves as the
intermediary between the application programs and the network operations.
4.25
• TCP creates a virtual connection between two communicating entities/TCPs to send

data. The receiver always sends either positive or negative acknowledgement about
the data packet to the sender, so that the sender always has the right clue about
whether the data packet has reached the destination or it needs to resend it.
• In addition, TCP uses flow and error control features to make TCP a reliable protocol.
4.4.1 TCP Services (Oct. 18)
• The services offered by TCP to the process at the application layer are explained
below:
1. Process-to-Process Communication:
• Like UDP, TCP provides process-to-process communication using port numbers.
• Port numbers are 16 bit long that help identify which process is sending or receiving
data on a host.
• Table 4.1 shows some well-known port numbers used by TCP.
Table 4.1: Well-known ports used by TCP
Port Protocol Description
7 Echo Echoes a received datagram back to the sender.
9 Discard Discards any datagram that is received.
11 Users Active users.
13 Day time Returns the date and time.
17 Quote Returns a quote of the day.
19 Chargen Returns a string of characters.
20 FTP, Data File Transfer Protocol (data connection).
21 FTP, Control File Transfer Protocol (control connection).
23 TELNET Terminal Network.
25 SMTP Simple Mail Transfer Protocol.
53 DNS Domain Name Server.
67 BOOTP Bootstrap Protocol.
80 HTTP Hypertext Transfer Protocol.
111 RPC Remote Procedure Call.
4.26
2. Stream Delivery Service: (Oct. 17)

• TCP is a stream oriented protocol. TCP allows the sending process to deliver data as a
stream of bytes and allows the receiving process to obtain data as a stream of bytes.
• TCP creates an environment in which the two processes seem to be connected by an
imaginary “tube”. This tube carries data across the Internet. This imaginary
environment is shown in Fig. 4.23.
• The sending process produces the stream of bytes and the receiving process reads data
from it.
Sending Receiving
process process
Stream of bytes
TCP TCP
Fig. 4.23: Stream Delivery
3. Sending and Receiving Buffers:

• TCP requires buffers for data storage, since the sending and receiving processes may
not write or read data at the same rate/speed.
• There are two buffers, sending buffer and receiving buffer, one for each.
Fig. 4.24: Sending and Receiving Buffers
4.27
• Fig. 4.24 shows the movement of the data in one direction. At the sending side, the
buffer is divided into three sections. The white section is empty, and can be filled by
the sending process.
• The cross-section area shows bytes are sent but not yet acknowledged. TCP keeps these
bytes in the buffer until it receives an acknowledgement. The dotted area contains
bytes to be sent by sending TCP.
• At receiver, the operation is simpler. The circular buffer is divided into two areas,
white and shaded.
• The white area contains empty buffers to be filled by bytes received from the network
layer.
• The shaded section contains received bytes that can be read by the receiving process.
When the byte is read, that part of the buffer becomes empty.
4. Segments:
• Fig. 4.25 shows segments in TCP.
• The IP layer, as a service provider for TCP, needs to send data in packets, not as a
stream of bytes. At the transport layer, TCP groups a number of bytes together into a
packet called a segment.
Fig. 4.25: TCP Segments
• TCP adds a header to each segment and delivers it to IP, for transmission. The
segments are encapsulated in IP datagram and transmitted.
5. Full Duplex Communication:
• TCP offers full duplex communication in which data can flow in both directions at the
same time.
4.28
6. Connection Oriented Service:

• TCP is a connection oriented protocol. When a process at site A wants to send and
receive data from another process at site B, the following occurs:
(i) The two TCPs establish a connection between them.
(ii) Data is exchanged in both directions.
(iii) The connection is terminated.
7. Reliable Service:
• TCP is connection oriented and reliable protocol. It uses acknowledgement to check
the arrival of data.
4.4.2 TCP Features (April 17, 19)

• To provide the services mentioned above, TCP has several features that are explained
below:
1. Numbering System:
• TCP software keeps track of segment (packets) transmitted and received. But there is
no number value in the segment header.
• There are two fields i.e., sequence number and the acknowledgement number. These
two fields refer to the byte number and not the segment number.
2. Byte Number:
• TCP numbers all data bytes that are transmitted in a connection.
• Numbering is independent in each direction. The numbering starts with a randomly
generated number.
3. Sequence Number:
• The value in the sequence number field of a segment defines the number of the first
byte contained in that segment.
• When a segment carries a combination of data and control information
(piggybacking), it uses a sequence number.
• If a segment does not carry user data, it does not logically define a sequence number.
• The field is there but value is not valid. Randomly generated sequence numbers are
used. If it is x then the first byte sequence number is x+1.
4. Acknowledgement Number:
• Communication in TCP is full duplex. Both communication parties send and receive
data at the same time.
• Every party starts with a different sequence number. Each party also uses an
acknowledgment number to confirm the bytes it has received.
• The value of the acknowledgement field in a segment defines the number of the next
byte a party expects to receive. The acknowledgement number is cumulative.
4.29
5. Flow Control:
• TCP provides a flow control mechanism. The receiver of data controls the amount of
data that is to be sent by the sender. By doing this, the receiver is not swamped by data
sent by the sender.
• The numbering system allows TCP to use a byte oriented flow control.
6. Error Control:
• For providing reliable service, TCP uses error control mechanisms. Error control is
byte oriented.
• TCP also provides congestion control. Receiver not only controls the amount of data
sent by the sender (flow control), but it is also determined by the level of congestion in
the network.
4.4.3 TCP Segment Format (April 18)
• A packet in TCP is called a segment. The format is shown in Fig. 4.26.
Header Data
Source port address Destination port address

16 bits 16 bits
Sequence number
32 bits
Acknowledgment number
32 bits
HLEN Reserved U A P R S F
Window size
4 bits 6 bits R C S S Y I
16 bits
G L H T N N
Checksum Urgent pointer
16 bits 16 bits
Options and Padding
Fig. 4.26: TCP Segment Format

• The segment consists of a 20 to 60 bytes header, followed by data from the application
layer.
• The header is of 20 bytes if no options are used and up to 60 bytes if it contains
options.
4.30
• Fig. 4.26 shows following fields of TCP segment format:

1. Source Port Address: This 16-bits field defines the port number of the
application program in the host that is sending the segment.
2. Destination Port Address: This 16-bits field defines the port number of the
application program in the host who is receiving the segment.
3. Sequence Number: This 32-bits field defines the number assigned to the first
byte of data contained in the segment.
4. Acknowledgement Number: This 32-bits field defines the byte number that the
receiver of the segment is expecting to receive from another party.
5. Header Length: This 4-bits field defines the length of the header. The length of
the header can range between 20 to 60 bytes.
6. Reserved: This 6-bits field is reserved for future use.
7. Control: This field defines 6 different control bits or flags as shown in Fig. 4.27.
One or more flags can be set at a time.
Note: URG : Urgent pointer is valid RST : Reset the connection
ACK : Acknowledgement is valid SYN : Synchronize sequence number
PSH : Request for push FIN : Terminate the connection
URG ACK PSH RST SYN FIN
Fig. 4.27: Control Fields

These bits enable flow control connection establishment and termination,
connection abortion, and the mode of data transfer in TCP.
The brief description of each bit is shown in Table 4.2.
Table 4.2: Description of flags in the control field
Flag Description
URG The value of the urgent pointer field is valid.
ACK The value of the acknowledgment field is valid.
PSH Push the data.
RST Reset the connection.
SYN Synchronize sequence number during connection.
FIN Terminate the connection.
8. Windows Size: This 16-bits field defines the size of the window in bytes that
another party must maintain. It is used for flow control between two stations and
4.31
indicates the amount of buffer (in bytes) the receiver has allocated for a segment,
i.e. how much data is the receiver expecting.
9. Checksum: This 16-bits field contains the checksum used for error control.
10. Urgent Pointer: This 16-bits field is valid only if the urgent flag is set. It is used
when the segment contains urgent data.
11. Option: There can be up to 40 bytes of optional information in the TCP header. It
facilitates additional options which are not covered by the regular header. Option
field is always described in 32-bit words. If this field contains data less than 32-bit,
padding is used to cover the remaining bits to reach 32-bit boundary.
4.4.4 TCP Connection Establishment and Termination

• TCP is a connection oriented protocol and every connection oriented protocol needs to
establish connection (a virtual path) in order to reserve resources at both the
communicating ends i.e., source and destination.
• In TCP, connection-oriented transmission requires three phases namely, connection
establishment, data transfer, and connection termination.
Connection Establishment:
• To establish a connection, TCP uses a three-way handshaking. Fig. 4.28 shows an
example of connection establishment using three-way handshaking.
• In this example we take an application program, known as the client, wants to make a
connection with another application program, known as the server, using TCP as the
transport layer protocol.
• The process starts with the server. The server program tells its TCP that it is ready to
accept a connection. This request is called a passive open.
• Although the server TCP is ready to accept a connection from any machine in the
world, it cannot make the connection itself.
• The client program issues or initiates a request for an active open. A client that wishes
to connect to an open server tells its TCP to connect to a particular server.
• TCP can now start the three-way handshaking process as shown in Fig. 4.28. To
establish a connection, the three-way (or 3-step) handshake occurs:
1. SYN Segment: In this segment only the SYN flag is set. SYN segment is for
synchronization of sequence numbers. The client in the example chooses a
random number as the first sequence number and sends this number to the
server. This sequence number is called the Initial Sequence Number (ISN). The
SYN segment is a control segment and carries no data. However, it consumes one
sequence number. When the data transfer starts, the ISN is incremented by 1. We
4.32
can say that the SYN segment carries no real data, but we can think of it as
containing one imaginary byte.
2. SYN+ACK Segment: It sets two flag bits set namely, SYN and ACK. A SYN + ACK
segment cannot carry data, but does consume one sequence number. SYN-ACK
segment has a dual purpose. First, it is a SYN segment for communication in the
other direction. The server uses this segment to initialize a sequence number for
numbering the bytes sent from the server to the client. The server also
acknowledges the receipt of the SYN segment from the client by setting the ACK
flag and displaying the next sequence number it expects to receive from the client.
Because it contains an acknowledgment, it also needs to define the receive window
size.
3. ACK Segment: ACK segment acknowledges the receipt of the second segment with
the ACK flag and acknowledgment number field. An ACK segment, if carrying no
data, consumes no sequence number.
Fig. 4.28: Connection Establishment of TCP using Three-way Handshaking
Data Transfer in TCP:

• After connection of TCP is established, bidirectional data transfer can take place. The
client and server can send data and acknowledgments in both directions.
• Fig. 4.29 shows an example of TCP in which, after a connection is established, the
client sends 2,000 bytes of data in two segments. The server then sends 2,000 bytes in
one segment.
4.33
• The client sends one more segment. The first three segments carry both data and
acknowledgment, but the last segment carries only an acknowledgment because there
is no more data to be sent.
• The data segments sent by the client have the PSH (push) flag set so that the server TCP
tries to deliver data to the server process as soon as they are received.
• The segment from the server, on the other hand, does not set the push flag. Most TCP
implementations have the option to set or not set this flag.
Fig. 4.29: Data Transfer in TCP

Connection Termination:
• Any of the two parties involved in exchanging/transferring data i.e., client or server
can close the connection, although it is usually initiated by the client.
4.34
• Most implementations today allow two options for connection termination for TCP
namely, three-way handshaking and four-way handshaking with a half-close option.
Three-Way Handshaking:
• Fig. 4.30 shows three-way handshaking for connection termination for TCP.
1. FIN Segment: In a common situation, the client TCP, after receiving a close
command from the client process, sends the first segment, a FIN segment in which
the FIN flag is set. The FIN segment consumes one sequence number if it does not
carry data.
2. FIN+ACK Segment: The server TCP, after receiving the FIN segment, informs its
process of the situation and sends the second segment, a FIN+ACK segment, to
confirm the receipt of the FIN segment from the client and at the same time to
announce the closing of the connection in the other direction. This segment can
also contain the last chunk of data from the server. If it does not carry data, it
consumes only one sequence number. The FIN + ACK segment consumes one
sequence number if it does not carry data.
3. ACK Segment: The client TCP sends the last segment, an ACK segment, to confirm
the receipt of the FIN segment from the TCP server. This segment contains the
acknowledgment number, which is one plus the sequence number received in the
FIN segment from the server. This segment cannot carry data and consumes no
sequence numbers.
Fig. 4.30
4.35
4.4.5 State Transition Diagram

• To keep track of all the different events happening during TCP connection
establishment, connection termination, and data transfer, TCP is specified as the finite
state machine shown in Fig. 4.31.
• The Fig. 4.31 shows the two FSMs (Finite State Machines (a mathematical model of
computation)) used by the TCP client and server combined in one diagram.
• In the Fig. 4.31, the ovals represent the states. The transition from one state to another
is shown using directed lines. Each line has two strings separated by a slash. The first
string is the input, which TCP receives. The second is the output, what TCP sends.
• The dotted black lines in the Fig. 4.31 represents the transition that a server normally
goes through; the solid black lines show the transitions that a client normally goes
through. However, in some situations, a server transitions through a solid line or a
client transitions through a dotted line.
• The dark black lines show special situations. Note that the oval marked as
ESTABLISHED is in fact two sets of states, a set for the client and another for the
server, that are used for flow and error control.
• Following table list of states for TCP:
Sr. No. State Description
1. CLOSED No connection exists.
2. LISTEN Passive open received; waiting for SYN.
3. SYN-SENT SYN sent; waiting for ACK.
4. SYN-RCVD SYN+ACK sent; waiting for ACK.
5. ESTABLISHED Connection established; data transfer in progress.
6. FIN-WAIT-1 First FIN sent; waiting for ACK.
7. FIN-WAIT-2 ACK to first FIN received; waiting for second FIN.
8. CLOSE-WAIT First FIN received, ACK sent; waiting for application to

close.
9. TIME-WAIT Second FIN received, ACK sent; waiting for 2MSL time-
out.
10. LAST-ACK Second FIN sent; waiting for ACK.
11. CLOSING Both sides decided to close simultaneously.
4.36
Fig. 4.31: State Transition Diagram of TCP
4.4.6 Windows in TCP

• TCP uses two windows (send window and receive window) for each direction of data
transfer, which means four windows for a bi-directional communication.
1. Send Window:
• The window we have used is of size 100 bytes (normally thousands of bytes), but later
we see that the send window size is dictated by the receiver (flow control) and the
congestion in the underlying network (congestion control).
4.37
(a) Send Window
(b) Opening, Closing and Shrinking Send Window
Fig. 4.32: Send Window in TCP
• The send window in TCP is similar to one used with the Selective Repeat (SR) protocol
but with some following differences:
o The nature of entities related to the window. The window in SR numbers pockets,
but the window in the TCP numbers bytes. Although actual transmission in TCP
occurs segment by segment, the variables that control the window are expressed
in bytes.
o The second difference is that, in some implementations, TCP can store data
received from the process and send them later, but we assume that the sending
TCP is capable of sending segments of data as soon as it receives them from its
process.
o Another difference is the number of timers. The theoretical SR protocol may use
several timers for each packet sent, but the TCP protocol uses only one timer.
2. Receive Window:
• The window we have used is of size 100 bytes (normally thousands of bytes). The
figure also shows how the receive window opens and closes; in practice, the window
should never shrink.
4.38
• There are two differences between the receive window in TCP and the one we used for
SR. These differences are given below:
o The first difference is that TCP allows the receiving process to pull data at its own
pace. This means that part of the allocated buffer at the receiver may be occupied
by bytes that have been received and acknowledged, but are waiting to be pulled
by the receiving process. The receive window size is then always smaller or equal
to the buffer size, as shown in the Fig. 4.33. The receiver window size determines
the number of bytes that the receive window can accept from the sender before
being overwhelmed (flow control). In other words, the receive window size,
normally called rwnd. It can be determined as rwnd = buffer size − number of
waiting bytes to be pulled.
o The way acknowledgments are used in the TCP protocol. Remember that an
acknowledgement in SR is selective, defining the uncorrupted packets that have
been received. The major acknowledgment mechanism in TCP is a cumulative
acknowledgment announcing the next expected byte to receive. The new versions
of TCP, however, uses both cumulative and selective acknowledgements.
(a) Receive Window and Allocated Buffer
(b) Opening and Closing of Receive Window
Fig. 4.33: Receive Window in TCP
4.39
Difference between TCP and UDP: (Oct. 18)

Sr.
Terms TCP UDP
No.
1. Acronym for Transmission Control User Datagram Protocol.
Protocol.
2. Connection TCP is a connection- UDP is a connectionless
oriented protocol. protocol.
3. Speed of transfer The speed for TCP is slower UDP is faster because there
than UDP. is no error-checking for
packets.
4. Header Size TCP header size is 20 bytes. UDP Header size is 8 bytes.
5. Weight TCP is heavy-weight. TCP UDP is lightweight. There is
requires three packets to no ordering of messages,
set up a socket connection, no tracking connections,
before any user data can be etc. It is a small transport
sent. TCP handles reliability layer designed on top of IP.
and congestion control.
6. Data Flow Control TCP does Flow Control. TCP UDP does not have an
requires three packets to option for flow control.
set up a socket connection,
before any user data can be
sent. TCP handles reliability
and congestion control.
7. Error Checking TCP does error checking UDP does error checking,
but no recovery options.
8. Reliability and Unreliable best-effort Reliable delivery of
Acknowledgements delivery without messages all data is
acknowledgements. acknowledged.
9. Retransmissions Not performed. Application Delivery of all data is
must detect lost data and managed, and lost data is
retransmit if needed. retransmitted
automatically.
10. Overhead Very low Low, but higher than UDP.
11. Data Quantity Small to moderate amounts Small to very large
Suitability of data. amounts of data.
4.40
PRACTICE QUESTIONS
1. The functions of transport layer of TCP/IP model include,
(a) Flow control (b) Congestion control
(c) Addressing (d) All of these
2. Which is an application program running on the host and uses the services of the
transport layer?
(a) Datagram (b) Packet
(c) Process (d) None of these
3. Which is the most common method to achieve process to process communication?
(a) Web paradigm (b) Client/server paradigm
(c) Programming paradigm (d) None of these
4. Which numbers are 16 bit long that help identify which process is sending or
receiving data on a host?
(a) Port (b) Packet
(c) Message (d) None of these
5. The transport layer address is called ______.
(a) Port address (b) Port number
6. The IANA (Internet Assigned Number Authority) has divided the port numbers into
three ranges,
(a) Well known ports (b) Registered ports
(c) Dynamic ports (d) All of these
7. Which is an address combination of IP address and a port number?
(a) Port address (b) Logical number
(c) Socket address (d) None of these
8. Which is a set of memory locations that can hold packets at the sender and
receiver?
(a) Buffer (b) Packet
(c) Message (d) All of these
9. Which is a service in the transport layer, observes that the data delivered to the
receiver is error free and reliable?
(a) Flow (b) Congestion
(c) Error (d) None of these
4.41
10. In which service, the packets are sent from one machine to another without
connection establishment.
(a) Connection-oriented (b) Connectionless
(c) Flow-oriented (d) All of these
11. Transport layer protocols include,
(a) TCP (b) SCTP
(c) UDP (d) All of these
12. Which is connectionless, unreliable transport layer protocol?
(a) TCP (b) SCTP
(c) UDP (d) All of these
13. UDP packets have a fixed-size header of ______ bytes.
(a) 16 (b) 8
(c) 32 (d) 64
14. Which is the most widely used protocol for data transmission in communication
networks as the Internet?
(a) IP (b) UDP
(c) TCP (d) SCTP
15. Services of TCP include,
(a) Flow control (b) Error control
(c) Congestion control (d) All of these
16. A packet in TCP is called ______.
(a) Segment (b) Datagram
(c) Header (d) None of these
17. Which is a connection oriented protocol?
(a) TCP (b) SCTP
(c) UDP (d) None of these
18. Which is a situation in which too many sources over a network attempt to send
data and the router buffers start overflowing due to which loss of packets occur?
(a) Congestion (b) Error
(c) Event (d) None of these
19. Stream Control Transmission Protocol (SCTP) combines the features of which
protocols?
(a) TCP (b) UDP
4.42
20. An endpoint of an inter-process communication flow across a computer network is

called ______.
(a) socket (b) pipe
(c) port (d) None of these
21. User datagram protocol is called connectionless because ______.
(a) All UDP packets are treated independently by transport layer
(b) It sends data as a stream of related packets
(c) both (a) and (b)
(d) None of these
22. In the sending computer, UDP receives a data unit from the ______ layer.
(a) Application (b) Transport
(c) Network (d) None of these
23. UDP uses ______ to handle outgoing user datagrams from multiple processes on one
host.
(a) flow control (b) multiplexing
(c) demultiplexing (d) None of these
24. The ports ranging from 49,152 to 65,535 can be used as temporary or private port
numbers. They are called the ______ ports.
(a) well-known (b) registered
(c) dynamic (d) None of these
25. A port address in UDP is ______ bits long.
(a) 8 (b) 16
(c) 32 (d) None of these
26. TCP uses ______ to check the safe and sound arrival of data.
(a) an acknowledgment mechanism (b) out-of-band signaling
(c) the services of another protocol (d) None of these
27. UDP packets are encapsulated in ______.
(a) an Ethernet frame (b) an TCP segment
(c) an IP datagram (d) None of these
28. TCP has ______; SCTP has ______.
(a) packets; segments (b) segments; packets
(c) segments; frames (d) None of these
29. UDP uses ______ to handle incoming user datagrams that go to different processes
on the same host.
(a) flow control (b) multiplexing
(c) demultiplexing (d) None of these
4.43
30. Communication in TCP is ______.

(a) simplex (b) half-duplex
(c) full-duplex (d) None of these
ANSWERS
1. (d) 2. (c) 3. (b) 4. (a) 5. (c) 6. (d) 7. (c)
8. (a) 9. (c) 10.(b) 11. (d) 12. (c) 13. (b) 14. (c)
15. (d) 16. (a) 17. (a) 18. (a) 19. (c) 20. (a) 21. (a)
22. (a) 23. (b) 24. (c) 25. (b) 26. (a) 27. (c) 28. (b)
29. (c) 30. (c)
1. The ______ layer is responsible for error-free, end-to-end delivery of data from the
source host to the destination host.
2. The transport layer provides a ______ control mechanism between the adjacent
layers of the TCP/IP model.
3. A ______ is identified for each transport protocol and address combination by a 16-
bits unsigned number, known as the port number.
4. A ______ is an application layer entity (running program) that uses the services of
the transport layer.
5. Encapsulation happens at the ______ site.
6. If the producer delivers the items after the consumer has requested them, the
delivery is referred to as ______.
7. Reliability can be achieved to add ______ control service to the transport layer.
8. When a packet is corrupted or lost, the receiving transport layer can somehow
inform the sending transport layer to resend that packet using the ______ number.
9. The ______ is represented as a set of slices, called the sliding window, that occupy
part of the circle at any time.
10. UDP is ______ protocol.
11. UDP packet called user ______.
12. TCP provides process-to-process communication using ______ numbers.
13. At the transport layer, TCP groups a number of bytes together into a packet called
a ______.
14. To establish a connection, TCP uses a ______ handshaking.
15. In UDP, ______ are associated with ports.
16. UDP is a simple, unreliable transport protocol, which does not provide ______ and
______ control.
4.44
17. The ______ is one of the most important protocols of the Internet Protocols suite
most widely used protocol for data transmission in communication networks such
as the internet.
18. User Datagram Protocol (UDP) is a ______ layer protocol.
19. UDP ______ and connectionless protocol.
20. TCP is a ______ layer protocol that provides for a connection-oriented, reliable
service to applications.
21. The combination of an IP address and a port number is called a ______ address.
22. A port number is a ______ address used to identify any client-server program
uniquely.
ANSWERS
1. transport 2. flow 3. port 4. process 5. sender
6. pulling 7. error 8. sequence 9. buffer 10. stateless
11. datagram 12. port 13. segment 14. three-way 15. queues
16. error, flow 17. TCP 18. transport 19. unreliable 20. transport
21. socket 22. 16-bits

1. The transport layer provides services such as connection-oriented
communication, reliability, flow control, and multiplexing.
2. TCP also prevents data loss due to a fast sender and slow receiver by imposing
some flow control techniques.
3. Error control uses open loop congestion control to prevent the congestion
and closed loop congestion control to remove the congestion in a network once it
occurred.
4. A process on the local host, called a client, needs services from a process usually on
the remote host called a server.
5. Decapsulation happens at the receiver site.
6. If the sender delivers items whenever they are produced without the prior request
from the consumer the delivery is referred to as pushing.
7. The error control service in transport layer, observes that the data delivered to the
receiver is error free.
9. In a connectionless oriented service, connection is established first and then data
are transferred in between sender and receiver. After the end of data transfer,
connection is released.
4.45
10. UDP provides a process to process communication using sockets, a combination of

IP addresses and port numbers.
11. UDP provides checksums for data integrity, and port numbers for addressing
different functions at the source and destination of the datagram.
12. UDP creates a virtual connection between two communicating entities/TCPs to
send data.
13. Sequence numbers are 16 bit long that help identify which process is sending or
receiving data on a host.
14. TCP is a stream oriented protocol. TCP allows the sending process to deliver data as
a stream of bytes and allows the receiving process to obtain data as a stream of
bytes.
15. TCP uses two windows (send window and receive window) for each direction of
data transfer.
16. In TCP, connection-oriented transmission requires three phases: connection
establishment, data transfer, and connection termination.
17. To send a message from one process to another, the TCP protocol encapsulates and
decapsulates messages in an IP datagram.
18. UDP services are Process to Process Communication, Connectionless Services, Flow
Control, Error Control, Congestion Control, Encapsulation and Decapsulation,
Queuing, Multiplexing and Demultiplexing.
19. UDP does not provide congestion control mechanisms.
20. Stream Control Transmission Protocol (SCTP) is a reliable, message-oriented
transport layer protocol.
21. UDP does not guarantee ordered delivery of data.
22. TCP unreliable and connectionless protocol.
23. SCTP provides some of the features of both UDP and TCP, (it is message-oriented
like UDP and ensures reliable, in-sequence transport of messages with congestion
control like TCP).
24. A connection-oriented protocol establishes a connection, manages the data
transfer and terminates the connection.
ANSWERS
1. (T) 2. (T) 3. (F) 4. (T) 5. (T) 6. (T)
7. (T) 8. (T) 9. (F) 10. (T) 11. (T) 12. (F)
13. (F) 14. (T) 15. (T) 16. (T) 17. (F) 18. (T)
19. (T) 20. (T) 21. (T) 22. (F) 23. (T) 24. (T)
4.46

1. List functions of the transport layer.
2. What is the port number?
3. What is a socket address?
4. Define pulling.
5. What is a buffer?
6. Give role of sequence number.
7. Define congestion.
8. List transport layer protocols.
9. UDP is connectionless protocol. State true or false.
10. List UDP services.
11. What is TCP?
12. Which services are provided by TCP?
1. Describe the process to process communication with diagrams.
2. Explain the following terms to transport layer:
(i) Multiplexing and demultiplexing. (ii) Encapsulation and decapsulation.
3. With the help of diagram describe pushing and pulling in the transport layer.
4. Write a short note on: Sliding window.
5. Describe connectionless and connection-oriented services in detail.
6. What are the features of UDP?
7. Explain datagram format of UDP.
8. What are the services provided by UDP? Explain two of them in detail.
9. With the help of example, describe checksum.
10. Which services are provided by TCP?
11. With the help of diagram describe segment format of TCP.
12. Explain data transfer process of TCP with diagram.
13. Compare TCP and UDP.
14. Write a short note on: Windows in TCP.
15. Describe state transition diagram of TCP in detail.

April 2016
1. Give header size of UDP packet. [1 M]
2. Write a short note on: UDP. [5 M]
4.47
April 2017
1. List any two features of TCP. [1 M]
2. Explain the concept of multiplexing and demultiplexing used in process to
process delivery. [4 M]
October 2017
1. State any two applications of UDP. [1 M]
2. Explain stream delivery service and sending and receiving buffer service
of TCP. [5 M]
April 2018
1. What is the window size of the TCP segment? [1 M]
2. Write a short note on: UDP. [5 M]
3. Explain any four features supported by TCP. [4 M]
October 2018
1. List the protocols used at the transport layer. [1 M]
2. List the services provided by TCP. [1 M]
3. Give difference between TCP and UDP. [5 M]
Ans. Refer to Page No. 4.40.
April 2019
1. Draw and explain UDP datagram. [5 M]
2. Explain TCP features. [1 M]
4.48
NOTES
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
❖❖❖
1
NOTES
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
______________________________________________________________________________________________
❖❖❖
2

Computer Networks I

Uploaded by

Copyright:

Available Formats

Computer Networks I

Uploaded by

Document Information

Original Title

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Computer Networks I

Uploaded by

Copyright:

Available Formats

A Book Of

Dr. Ms. Manisha Bharambe Mrs. Veena K. Gandhi

Published By : Polyplate Printed By :

We take an opportunity to present this Text Book on "Computer Networks-I" to the

1. Introduction to Networks and Network Models 1.1 – 1.82

2. Lower Layers 2.1 – 2.98

3. Network Layer 3.1 – 3.90

4. Transport Layer 4.1 – 4.48

1.1 DATA COMMUNICATION

• Data communication is the exchange of information between two computers capable

Fig. 1.1: Communication between Two Computers

1.1.1 Definition of Data Communication

• Fig. 1.2 shows the following components of data communication:

1.1.3 Data Representation

3. Images: Images are also represented by bit patterns. An image is composed of a

1.1.4 Data Flow (Oct. 17)

Fig. 1.3: Simplex Communication Mode

(ii) Cheaper in cost.

Direction of data at time 1

Direction of data at time 2

Fig. 1.4: Half-duplex Communication Mode

Advantages Half Duplex Communication Mode:

Direction of data all the time 1

1.2.1 Computer Networks (April 17, 19)

• A computer network is an interconnection of computers and computing devices using

Fig. 1.6: Network Nodes and Links

Fig. 1.7: Computer Network (Wired and Wireless)

• A wired network is simply, a collection of nodes connected by cables like Ethernet,

1.2.2 Definition of Computer Network

• Networking plays an important role in every kind of organization from small to

Fig. 1.8: Typical Computer Network

1.2.3 Network Criteria

1.2.4 Physical Structures

1.2.4.1 Type of Connection

Fig. 1.9: Point-to-point Connection

Fig. 1.10: Broadcast Connection

1.2.4.2 Physical Topology

1.2.4.2.1 Star Topology (April 17, 18, 19, Oct. 18)

• In star topology each device has a dedicated point-to-point link on it to a central

• Communication is controlled by a central controller (Hub/Switch) only. Star topology

Fig. 1.11: Star Topology

1.2.4.2.2 Bus Topology (April 16)

Fig. 1.12: Bus Topology

1.2.4.2.3 Ring Topology (April 18, Oct. 17)

Fig. 1.13: Ring Topology

Host existence Depends on network Depends on network Yes.

1.2.4.2.4 Mesh Topology (April 16, 17, 18, 19)

(a) Full Mesh Topology (b) Partial Mesh Topology

Fig. 1.14: Mesh Topology

1.2.4.2.5 Tree Topology

5. No disruptions to the network then connecting or removing devices.

Fig. 1.15: Tree Topology

1.2.4.2.6 Hybrid Topology

• Fig. 1.16 shows an example of star-bus hybrid topology.

Fig. 1.16: Hybrid Topology

1.2.5 Network Types

Fig. 1.17: Types of Computer Network

Fig. 1.18: Geographical Arrangement of LAN, WAN and MAN

1.2.5.1 Local Area Network (LAN)